Method and system for processing telephone calls via a remote tie-line

ABSTRACT

A method and system include routing telephone calls from multiple private networks via public lines and trunks in a public switched telephone network (PSTN) to a host switch, at which the connections would begin to run parallel. The host switch routes the telephone calls via a private trunk group to a private facility, such as an electronic tandem network (ETN) node or an asynchronous transfer mode (ATM) switch. If the private trunk group is full or unavailable, the host switch routes the telephone calls over alternate private or public trunk groups and selected carriers.

CROSS-REFERENCE TO RELATED APPLICATIONS

[0001] This application hereby incorporates by reference in theirentireties the disclosures of the following applications, filedconcurrently herewith: “Dialing Plan Service Including Outgoing CallScreening, Remote Tie-Line Routing, Call Data Reporting andBilling”(attorney docket no. P20142), “Outgoing Call Screening”(attorneydocket no. P20143), “Method and System for Generating Call DataReports”(attorney docket no. P20344) and “Billing for AbbreviatedDialing Plan Service”(attorney docket no. P20345).

BACKGROUND OF THE INVENTION

[0002] 1. Field of the Invention

[0003] The present invention relates to the field of telecommunications.More particularly, the present invention relates to area wide centralexchange service (centrex) and private branch exchange (PBX) systems.

[0004] 2. Acronyms

[0005] The written description provided herein contains acronyms thatrefer to various telecommunications services, components and techniques,as well as features relating to the present invention. Although some ofthese acronyms are known, use of these acronyms is not strictlystandardized in the art. For purposes of the written description herein,the acronyms are defined as follows:

[0006] Access Code (AC)

[0007] Advanced Intelligent Network (AIN)

[0008] American Standard Code for Information Interchange (ASCII)

[0009] Automatic Selection of Facilities-Remote Tie Line (ASF-RTL)

[0010] Asynchronous Transfer Mode (ATM)

[0011] Called Party Number (CDN)

[0012] Calling Party Number (CPN)

[0013] Call Type Code (CTC)

[0014] Carrier Identification Code (CIC)

[0015] Central Exchange Service (Centrex)

[0016] Centralized Route Selection (CRS)

[0017] CentrexSMART Front End (CFE)

[0018] Custom Virtual Network (CVN)

[0019] Customized Dialing Plan-Access Code (CDP-AC)

[0020] Direct Inward Dial (DID)

[0021] Electronic Tandem Network (ETN)

[0022] File Transfer Protocol (FTP)

[0023] Initial Address Message (IAM)

[0024] Integrated Service Control Point (ISCP)

[0025] Interexchange Carrier (IXC)

[0026] Local Access and Transport Area (LATA)

[0027] Local Exchange Carrier (LEC)

[0028] Lucent Service Control Point (LSCP)

[0029] Manipulation Dialing Plan (MDP)

[0030] Multi-Frequency (MF)

[0031] Numbering Plan Area (NPA) a.k.a. area code

[0032] North American Numbering Plan (NANP)

[0033] Nature Of Number (NON)

[0034] Off Hook Delay (OHD)

[0035] Original Called Number (OCN),

[0036] Outgoing Call Screening (OCS)

[0037] Point of Presence (POP)

[0038] Private Network (PN)

[0039] Privilege Class (PC)

[0040] Private Branch Exchange (PBX)

[0041] Private Numbering Plan (RXX)

[0042] Public Office Dial Plan (PODP)

[0043] Public Switched Telephone Network (PSTN)

[0044] Redirected Number (RDN)

[0045] Regional Bell Operating Company (RBOC)

[0046] Remote Tie Line Billing Reduction (RBR)

[0047] Service Control Point (SCP)

[0048] Service Switching Point (SSP)

[0049] Signaling System 7 (SS7)

[0050] Signaling Transfer Point (STP)

[0051] Station Message Detail Recording (SMDR)

[0052] Special Dialing Plan (SDP)

[0053] Terminating Attempt Trigger (TAT)

[0054] Transaction Capabilities Application Part (TCAP)

[0055] Transmission Control Protocol/Internet Protocol (TCP/IP)

[0056] Trunk Group (TG)

[0057] Usage Billing Reduction (UBR)

[0058] Usage Billing Suppression (UBS)

[0059] Virtual Private Network (VPN)

[0060] 3. Background Information

[0061] Currently, enterprises are removing private electronic tandemnetworks (ETNs), which are dependent on a network of private lines, inorder to reduce network expenses. With the reduction in prices on publicswitched telephone network (PSTN) usage and the increase in price ofprivate lines, it is desirable for enterprises to direct their calltraffic over the PSTN using a local Regional Bell Operating Company(RBOC) virtual network, if available. Increased use of the PSTN alsoreduces responsibility for telecommunications maintenance functionswithin the enterprises' domain, which are inherent with any systemdependent on private facilities, such as ETN and asynchronous transfermode (ATM) facilities. Although many enterprises are comfortable sendingas much traffic as possible over the PSTN, they may desire to retainportions of their private facilities, such as ATM backbones andinternational links, as needed, thus creating the need for a hybridvirtual private network-private network (VPN-PN).

[0062] The virtual network market is growing rapidly. Virtual networkofferings by the interexchange carriers (IXCs) such as AT&T's SoftwareDefined Network (SDN), MCI's Virtual Networking Service (VNET), andSprint's Virtual Private Network (VPN) are the primary market players invirtual network services.

[0063] AT&T markets SDN primarily to large users and has been one of themost popular services offered by resellers and aggregators. AT&T alsooffers Software Defined Data Network (SDDN) as an optional feature. SDDNprovides data networking capability for its SDN service. SDN providesusers with on-line network management capabilities to proactivelymonitor and control performance, security, accounting, network planningand configuration which is a major strength. AT&T has improved itsbilling features by introducing OneNet, a new SDN billing service.OneNet combines the charges from a customer's SDN and 800 services ontoone bill. OneNet does not, however, provide a predictable monthly bill.

[0064] MCI's VNET provides customers with long distance, voice, data andmessaging services, both on domestic and international levels. VNETsupports voice and data transmission up to 64 Kbps. VNET also providesoptional features such as an integrated network management service. Amajor strength of VNET is the centralized software defined database,which can be controlled directly from a workstation on the customer'spremises. MCI also offers MCI Perspective, which is a PC based softwareanalysis tool allowing customers to track, analyze and control theirtelecommunications billing information. MCI Perspective does not,however, provide a predictable monthly bill.

[0065] Sprint's VPN is a voice and data network operated by a singlesoftware controlled management system through the use of sharedtransmission facilities. VPN supports 56 Kbps transmission and wasdesigned for large corporate telecommunications users with multiplelocations. Sprint's Insight Executive integrates the network managementof VPN with other Sprint services such as 800, WATS, etc. onto a singleplatform. Another strength is Sprint's Insite ACT, a phone-based servicechange tool that allows VPN customers to add, change or cancel their VPNservice.

[0066] In FLEISCHER, III et al., U.S. Pat. No. 5,974,133, the disclosureof which is expressly incorporated herein by reference in its entirety,an overlay method and system are described for a multiple locationcommunications network, which provide additional telecommunicationservices, such as abbreviated dialing plans, automatic selection ofrouting, centralized access to private and public network facilities,and outgoing call screening. FLEISCHER, III et al.'s system is a goodbasic system, but provides limited flexibility and offers limitedoptions for automatic selection of routing, centralized access toprivate network facilities and outgoing call screening. FLEISCHER, IIIet al.'s system also does not provide predictable periodic billing orcentralized data collection.

[0067] Routing traffic to hub switches is well known in the field oftelecommunications. For example, MOSS et al., U.S. Pat. No. 5,917,899,the disclosure of which is expressly incorporated herein by reference inits entirety, teach a method for providing an interLATA virtual privatenetwork, using host or “hub” switches in the PSTN to connect callswithin the network. An interLATA, intra-network call is routed, based oninstructions from an SCP, from an originating switch to a first hubswitch within the same LATA as the originating switch and then to asecond hub switch within the same LATA as the called party number. MOSSet al.'s system is limited, though, to accommodating abbreviated dialingfor in-network calls across more than one LATA. Also, MADOCH et al.,U.S. Pat. No. 5,995,605, the disclosure of which is expresslyincorporated herein by reference in its entirety, teach a hub switchthat combines calls from multiple switches to a centrex telephone lineinto a data stream. The data stream is transmitted over a digital trunkto an information network node servicing a high traffic computernetwork, such as an Internet access system. MADOCH et al.'s system islimited to routing calls to an information network having apredetermined access number and does not handle routine customertelephone traffic.

[0068] The overarching need is to have a comprehensive networkalternative to private facilities, such as ETN, including centralizedadministration of carrier and private route selections as well ascentralized call screening functions. Moreover, the “virtual network”should be a uniform application. Currently, the virtual network systemsdo not provide the needed flexibility for efficiently merging a PSTNsystem and a private network to maximize call flow efficiency and costcontrol.

[0069] Rather, the systems either restrict calls to the PSTN or toprivate networks with little flexibility, or the systems route callsover a combination of the PSTN and private networks, based on a realtime analysis of trunk and carrier availability, often resulting induplication of effort or overuse of one of the sets of resources. Forexample, small remote locations within a private network tend to funnelcalls, as a practical matter, through the same ETN node. Use of onlyprivate trunk groups in this situation is overly expensive because thecall traffic from individual locations is insufficient to warrant theexpense of the private lines. Use of only the PSTN therefore appears tobe more appropriate. However, at some point, the aggregated calls fromseveral remote locations begin to parallel one another to access thesingle ETN node, thus overusing the PSTN and causing excessive expenseto the customer. A private trunk group dedicated to the ETN node in thissituation would handle the call traffic less expensively.

[0070] Conventional systems also offer limited reporting functions. Forexample, current systems produce call data records, referred to asstation message detail recording (SMDR), which is used by customers todetermine outgoing call activity on a station by station basis. SMDRcall details are accessible through an optional SMDR port on a PBX wherethe call originates, or in the case of centrex, through the servingnetwork central office switch where the call originates. With centrex,SMDR records flow to an SMDR host central processing unit in preparationfor SMDR delivery to the customer. The centrex SMDR network is known ascentrex station message detail recording and transmission(CentrexSMART). The current centrex and PBX SMDR systems, however, areindependent of one another so it is not possible for the customer withboth PBX and centrex SMDR records to have the data streams combined anddelivered to the customer as one aggregated data stream direct from theserving PBX or network switch.

[0071] Furthermore, the collection and dissemination of the SMDR data iscurrently very decentralized. The SMDR data on which the calling recordsare based are collected at local service switching points. Therefore,each PBX providing SMDR data requires a specific connection to its SMDRport, and in the case of centrex, each network switch must have directconnectivity to the SMDR host (i.e., CentrexSMART) for the data to betransferred. Also, each switch/PBX must be loaded with a relativelyexpensive SMDR package, for instructions regarding records to captureand transmit. For example, if the customer's network is serviced by fiveservice switching points, the customer or local exchange carrier musthave five SMDR packages loaded at all five locations.

BRIEF DESCRIPTION OF THE DRAWINGS

[0072] The present invention is further described in the detaileddescription that follows, by reference to the noted plurality ofdrawings by way of non-limiting examples of embodiments of the presentinvention, in which like reference numerals represent similar partsthroughout several views of the drawings, and in which:

[0073]FIG. 1 is a block diagram showing an exemplary telecommunicationsnetwork, according to an aspect of the present invention;

[0074]FIG. 1a is a block diagram showing a switch focusedtelecommunications network architecture, according to an aspect of thepresent invention;

[0075]FIG. 2 is a flow chart showing an exemplary common service logic,according to an aspect of the present invention;

[0076]FIG. 3 is a flow chart showing an exemplary privilege classservice logic, according to an aspect of the present invention;

[0077]FIG. 4 is a block diagram showing an exemplary telecommunicationsnetwork for implementing an automatic selection of facilities-remote tieline (ASF-RTL) feature, according to an aspect of the present invention;

[0078]FIG. 5 is an exemplary flow chart of the ASF-RTL service logic,according to an aspect of the present invention;

[0079]FIG. 6 is an exemplary flow chart of the ASF-RTL routing servicelogic, according to an aspect of the present invention;

[0080]FIG. 7 is an exemplary call flow diagram showing routing of atelephone call through the ASF-RTL system, according to an aspect of thepresent invention;

[0081]FIG. 8 is an exemplary call flow diagram showing routing of atelephone call through the ASF-RTL system to an international gateway,according to an aspect of the present invention;

[0082]FIG. 9 is a flow chart showing an exemplary PSTN billing servicelogic, according to an aspect of the present invention;

[0083]FIG. 10a is a block diagram showing an exemplarytelecommunications network for implementing a Virtual SMDR system,according to an aspect of the present invention;

[0084]FIG. 10b is a block diagram showing an alternate exemplarytelecommunications network for implementing a Virtual SMDR system,according to an aspect of the present invention; and

[0085]FIG. 11 is an exemplary flow chart of the Virtual SMDR servicelogic, according to an aspect of the present invention.

DETAILED DESCRIPTION OF EMBODIMENTS

[0086] In view of the above, the present invention through one or moreof its various aspects and/or embodiments is presented to accomplish oneor more objectives and advantages, such as those noted below.

[0087] An aspect of the present invention provides a method forefficiently routing a call from a private telecommunications networkthrough a public switched telephone network (PSTN) and multiple trunkgroups, including at least one private trunk group. The privatetelecommunications network may include both central exchange service(centrex) and private branch exchange (PBX) systems. The method includessetting a called party ID to correspond to an automatic selection offacilities-remote tie-line (ASF-RTL) telephone number and routing thecall through a public trunk group from a first switch to a switchhosting the ASF-RTL telephone number. The call is then routed through aprivate trunk group from the switch hosting the ASF-RTL telephone numberto a private customer facility hosting the called party number. Themethod may include initially determining whether the call is subject toASF-RTL processing.

[0088] In an embodiment of the invention, the method further includesrouting the call from the switch hosting the ASF-RTL telephone number tothe private customer facility hosting the called party number through analternative trunk group whenever the private trunk group is full orunavailable. Furthermore, whenever a service control point (SCP)determines that a called party number of the call is an internationalnumber, the call is routed from the private customer facility to aninternational gateway.

[0089] Another aspect of the present invention provides a method forefficiently routing a call to a called party number from a privatetelecommunications network, which includes at least one centrex or PBX,through a PSTN and multiple trunk groups. The call routing methodincludes querying an SCP for call processing instructions for the calland identifying whether the call is subject to ASF-RTL processing. Thequery includes a calling party ID corresponding to a calling partynumber and a called party ID corresponding to the called party number.When the call is subject to ASF-RTL processing, the called party ID isset to correspond to an ASF-RTL telephone number and an original calledparty ID is set to correspond to the called party number. Callprocessing instructions are then transmitted from the SCP to anoriginating service switching point. The call processing instructionsinclude the calling party ID, the called party ID and the originalcalled party ID. The call is routed to a host service switching pointthat corresponds to the called party ID.

[0090] The SCP is queried for additional call processing instructions.The query includes the calling party ID, the called party ID and theoriginal called party ID. Additional call processing instructions aredetermined, which include at least one private trunk group through whichto route the call. The called party ID is reset to correspond to theoriginal called party ID. The additional call processing instructionsare transmitted from the SCP to the host service switching point and thecall is routed from the host service switching point to a privateswitching facility of the customer through the private trunk group. Theprivate switching facility may include electronic tandem network nodesand asynchronous transfer mode switches. The method may also includerouting the call to a destination service switching point and connectingthe calling party number with the called party number.

[0091] In another aspect of the invention, the additional callprocessing instructions are determined by first determining at least onealternative trunk group and at least one carrier by which to route thecall. Routing the call from the host service switching point thenincludes determining whether the private trunk group is full orunavailable and, whenever the private trunk group is full orunavailable, routing the call to the private switching facility by analternative trunk group. When the alternative trunk group is a publictrunk group, the routing also includes a carrier.

[0092] In an embodiment of the invention, determining whether thetelephone call from the calling party number to the called party numberis subject to ASF-RTL is based on the type of call. Other embodimentsdetermine whether the telephone call from the calling party number tothe called party number is subject to ASF-RTL processing based on thedate, the day of week, the time of day or a percentage allocation.

[0093] Another aspect of the present invention provides a system forefficiently routing a call from a private telecommunications networkthrough a PSTN and a plurality of trunk groups, including at least oneprivate trunk group associated with an ASF-RTL telephone number. Theprivate telecommunications network includes a centrex, a PBX, or both.The system includes an SCP, which processes telephone calls, andmultiple service switching points. A first service switching pointqueries the SCP in response to the call from the privatetelecommunications network. A second service switching point includes ahost switching point for the private trunk associated with the ASF-RTLtelephone number and a third service switching point includes a privatefacility of the customer. The SCP determines from the query whether thecall is subject to routing through the second switching point. If so,the SCP instructs the first service switching point to route the call tothe second service switching point. The second service switching pointthen routes the call to the third service switching point through theprivate trunk group associated with the ASF-RTL telephone number.

[0094] In another aspect of the invention, the system also includes aninternational gateway. When the SCP determines that the called partynumber is an international telephone number, the call is routed from thethird service switching point to the international gateway. Likewise,the system may include an interexchange carrier. When the SCP determinesthat the called party number is a long distance telephone number, thecall is routed from the third service switching point to theinterexchange carrier.

[0095] Another aspect of the present invention provides a system forefficiently routing a call from a private telecommunications network ofa customer, including a centrex or a PBX, through a PSTN and multipletrunk groups, including at least one private trunk group. The systemincludes an SCP, which processes telephone calls and has an ASF-RTLrouting database. The system also includes an originating serviceswitching point, which launches a trigger to the SCP in response to acall initiated at a telephone in the customer's privatetelecommunications network, and a host switching facility, which hostsan ASF-RTL telephone number associated with the private trunk group. Thehost switching facility receives the call from the originating serviceswitching point based on instructions from the SCP. The system alsoincludes a private switching facility that hosts at least one secondprivate telecommunications network and connects to the host switchingfacility by the private trunk group. The private switching facility,which can include electronic tandem network nodes and asynchronoustransfer mode switches, receives the call from the host switchingfacility based on routing instructions from the SCP. The system may alsoinclude a terminating service switching point to which the call isdirected from the private switching facility. The terminating serviceswitching point is instructed by the SCP to terminate the call at thecalled party number.

[0096] The SCP receives a calling party ID and a called party IDcorresponding to the call from the originating service switching pointand determines from the ASF-RTL routing database whether the call issubject to ASF-RTL routing. If the call is subject to ASF-RTL routing,the SCP sets the called party ID to be the ASF-RTL telephone numberassociated with the private trunk group, so that the originating serviceswitching point forwards the call to the host switching facility. Therouting instructions include instructions on transmitting the call fromthe host switching facility to the private switching facility by theprivate trunk group.

[0097] In alternative embodiments of the invention, the ASF-RTL routingdatabase includes routing information based on any combination of thetype of call, the date, the day of week, the time of day and apercentage allocation. The routing instructions may also include themultiple trunk groups, which, in addition to the private trunk group,include alternative trunk groups and carriers. When the private trunkgroup is not available or full, the host switching facility selects oneof the alternative trunk groups and, if the selected alternative trunkgroup is a public trunk group, one of the carriers by which to forwardthe call to the private switching facility. Each of the multiple trunkgroups includes multi-frequency trunks.

[0098] This present invention upgrades the current Custom VirtualNetwork (CVN) application, increasing functionality of the application.At this time CVN is tariffed in California under Advice Letter(s) 18399and 17689. CVN is a set of features that improves the networkingcapabilities of customers connected to a telecommunications carrier'scentral offices. CVN is targeted toward multi-location/multi-switchsubscribers with a requirement for internal network calling. Theexisting CVN product is designed for centrex and PBX applications.

[0099] The present invention relates to an improvement of CVN. The newservice provides connectivity between multiple customer locationsthrough the use of the PSTN instead of through a private network.Declining costs for usage of the PSTN and rising costs for privatenetworks make CVN a desirable solution. However, the new servicerecognizes that there are some high volume situations in which use ofprivate trunk groups and/or use of specific interexchange carriers(IXCs) for traffic across local access and transport areas (LATAs) isstill preferable.

[0100] Among the main upgrades is an automatic selection offacilities-remote tie line (ASF-RTL) call routing system that routestelephone calls from a centrex or PBX system through the PSTN to acentral hub that routes the calls over private facilities to accommodatean anticipated high volume of call traffic. The ASF-RTL call routingsystem enhances efficiency, especially for calls requiring specialtreatment, such as those passing through an international gateway or forcalls from network locations having a relatively low volume of calltraffic. The ASF-RTL system also provides alternative routing via thePSTN to accommodate overflow traffic from private trunk groups (i.e.,tie lines), as necessary. The system simply and efficiently resolves theproblems with current call routing.

[0101] Another main upgrade is usage billing treatment, which includesusage billing suppression (UBS) and usage billing reduction (UBR). Usagebilling suppression determines whether a call is eligible for the UBSservice and if so, zero rates usage on centrex intraLATA voice usage andcharges this usage on a flat rate basis as a service feature charge.Usage billing reduction is available on PBX as well as on ASF-RTL calls.This feature allows flexible rating on intraLATA voice usage.

[0102] Another upgrade is outgoing call screening (OCS) based onprivilege classes (PC). The present invention also permits datacollected from the Advance Intelligent Network (AIN) service controlpoint (SCP) to be made available to the company in the an SMDR format.

[0103] The present invention relates to a uniform/simplified dialingplan that allows customers to create their own abbreviated networkdialing plan. The present invention also replaces current ETN dialingpatterns and phone numbers. In addition, it adds features andfunctionality including: creating centralized ASF-RTL hub sites foraggregation of traffic to point of presence ( POP) carrier connectionsor private backbone facilities and establishing privilege class levelsfor each outgoing call type (outgoing call screening (OCS)).

[0104] According to an aspect of the present invention, three differentusage billing related options are provided. Centrex to centrex callsthat are intraLATA voice calls are eligible for the usage billingsuppression (UBS) option. According to usage billing suppression, whenboth the calling station and the called station are determined to becentrex stations, and the call is an intraLATA voice call, a uniquebilling record is generated. A billing system will detect the uniquebilling record, causing the record to be discarded.

[0105] PBX to PBX calls, PBX to centrex calls, and centrex to PBX calls,that are intraLATA voice calls are eligible for the usage billingreduction (UBR) option. According to usage billing reduction, when acall is between fully participating UBR provisioned PBX telephonenumbers and UBS centrex telephone numbers, a unique billing record isgenerated. The billing system detects the unique billing record, causingthe billing record to be adjusted according to specific customercontract terms. The adjustment can be from zero to 100% of the standardrate for similar calls.

[0106] The billing system also detects intraLATA voice calls routed toASF-RTL centralized hub sites through a unique billing record. Thedetected billing record is then priced, according to specific customercontract terms, from zero to 100% of the standard price for such a call.This billing option will be referred to as RBR.

[0107] It should be noted and understood that the abbreviated dial planof the present invention is a service in addition to the customer'sexisting service, i.e., centrex, PBX, etc. Because the abbreviated dialplan is an overlay, the customer must decide how many of their centrexstations or PBX trunks they want equipped with the additional featuresof the present invention.

[0108] An advantage of the present invention is that it centralizes anenterprise's dialing pattern choices, their routing choices and theircall screening configurations. When the customer subscribes to eachfunction, the customer will experience ease of dialing, selectiverouting, network optimization, outbound dialing disaster recovery forcarriers or chosen routes as well as centralized management of outgoingcall screening.

[0109] The present invention operates in an AIN environment, includingswitches such as 5ESS switches manufactured by Lucent Technologies,Inc., DMS- 100 switches manufactured by Nortel Networks Corporation(Nortel), AXE-10 switches manufactured by Telefonaktiebolaget LMEricsson, or EWSD switches available from Siemens Information andCommunication Networks, Inc. The 5ESS switches may utilize an AINRelease 0.1 protocol and should be equipped with Generic 5E12 (orhigher) software and associated AIN SSP features. The DMS-100 switches(release NA009) may utilize an AIN Release 0.1 protocol and associatedAIN SSP features. Similarly, AXE-10 switches and EWSD switches mayutilize an AIN Release 0.1 protocol.

[0110] An exemplary network in which the present invention may operateis now discussed with respect to FIG. 1. FIG. 1 shows a hybrid VPN-PNnetwork architecture, but does not show the PSTN. Although the PSTNinterconnects all centrexs and PBXs, the PSTN connections are not shown.All the trunk connections shown are private trunk connections.

[0111]FIG. 1 shows locations AA and AB, which use the present inventionto route originating 7+ or 9+ dialed traffic over the PSTN. Location Ahas a dedicated private trunk group to one of the asynchronous transfermode (ATM) switches. This connection is necessary because, for example,location A is very large, so the aggregated traffic is cheaper to moveover a private trunk group to the ATM switch and the private networkthan it is over the PSTN. Location A is not using the functionality ofthe present invention.

[0112] Locations BA, and BB employ the features of the present inventionto route the originating 7+ or 9+ dialed traffic over the PSTN. FIG. 1shows location B having a dedicated private trunk group to one of theelectronic tandem network (ETN) nodes. Similar to location A, the use ofa private trunk group in conjunction with the ETN node is advantageousbecause location B is very large and at least a portion of the trafficis less expensive to move over a private trunk group to the ETN node andthe private network than it is over the PSTN. Unlike location A,however, location B employs features of the present invention because atleast a portion of the traffic is cheaper to move over the PSTN than itis over a private trunk group to the ETN node and the private network.

[0113]FIG. 1 also shows locations CA and CB, which have private trunksto location C. Thus, the customer is now using features of the presentinvention at locations CA and CB to route the originating 7+ or 9+dialed traffic, but private trunks are still being used. Location C hasa dedicated private tk group to an ATM switch. This is because locationC is very large and at least a portion of the traffic is cheaper to moveover a private trunk group to an ATM switch and the private network thanit is over the PSTN.

[0114]FIG. 1 shows an ASF-RTL hub near locations D, DA, and DB. ThisASF-RTL hub provides locations D, DA, and DB with direct access to aninternational gateway IG and to an ATM switch. Note that locations D,DA, and DB do not include interconnecting private trunks becauselocations D, DA, and DB are relatively small and do not justify theexpense of dedicated private trunk groups, as in the case of C, CA, andCB, for example. Therefore, traffic from locations D, DA, and DB ridesPSTN trunks to the ASF-RTL hub, at which point the traffic is aggregatedand routed over a private trunk group to the international gateway IG orthe ATM switch, depending on the called party number. Furthermore, theASF-RTL hub is tied to an ATM switch associated with locations C, CB,and CA and locations A, AA, and AB, providing ASF-RTL services to callsoriginating in these locations as well.

[0115] The service switching points (SSPs) 10, 12, 14 supporting thenetwork of FIG. 1 are shown in FIG. 1a. Triggers T are set in theswitches 10, 12 for subscribing stations AA, AB, BA, BB. The stationsEA, EB connected to switch 14 (and associated triggers T) are describedbelow. In one embodiment, customized dialing plan-access code (CDP-AC)triggers are employed for all centrex stations. Most PBX stations,except for PBX stations having PBX trunks that are trunk side multifrequency trunks one way outgoing from the PBX, which do not send accesscodes, and except for PRI trunks, also use a CDP-AC trigger. The PBXstations having one way trunks use off hook delay (OHD) triggers. ThePRI stations use PRIBC triggers. All triggers are based on originatingtelephone numbers.

[0116] The switches 10, 12 launch queries to a service control point(SCP) 20 via a local STP 16 and a regional STP 18. The SCP 20 translatesabbreviated dialed numbers into numbers understood by the PSTN. Thesetranslations are highly centralized in that the translations only happenat the SCP 20, but because the responses are directed to the queryingSSPs 10, 12, 14, the effect is highly decentralized: it is as if theSSPs 10, 12, 14 are doing the translations. The SSPs 10, 12, 14 alsolaunch queries to the SCP 20 to effect other vertical features, such asASF-RTL and UBS. Terminating only (TO) telephone numbers and Off-nettelephone numbers are also shown in FIG. 1a. Off-net telephone numberswill be discussed below.

[0117] A terminating only telephone number is a telephone number thatwill not cause a trigger, thus preventing the end user of theterminating only telephone number from using a dialing plan. Terminatingonly telephone numbers, however, are listed in dialing plan tables.Consequently, the customer can prevent some triggering end users fromreaching certain terminating only telephone numbers, while allowingother triggering end users to reaching those same terminating onlytelephone numbers. The control can be provided through outgoing callscreening, described below. Terminating only telephone numbers can befor either customer controlled stations (which do not trigger) orstations not controlled by the customer, e.g., suppliers.

[0118] By way of example, the SCP 20 is implemented with a BellcoreIntegrated Service Control Point and loaded with ISCP software Version4.4 (or higher), available from Telecordia, Murray Hill, N.J. In analternative embodiment of the invention, the SCP 20 may be a LucentAdvantage SCP, with software release 94, available from LucentTechnologies, Inc.

[0119] In the network of FIGS. 1 and 1a, some abbreviated dialed numbersare translated into long distance numbers that are routed to the IXCpoint of presence M over the PSTN. Thus, the calls are routed by the IXCswitch M to the destination. Similarly, international calls are routedto the international gateway IG for routing to the internationaldestination. Although a single IXC POP and single international gatewayare shown, additional international gateways and IXC POPs can beprovided, if desired.

[0120] All private trunks are multi-frequency (MF) and, hence, do notsend the calling party number (CPN). However, the calls going over thePSTN have a high degree of likelihood of being carried end to end viaSS7, and, hence, the calls deliver the calling party number if the CDNhas Caller ID. That is, the trunks between end offices 10, 12, 14 areSS7 trunks. The links between the switches 10, 12, 14, M are SS7 links.

[0121] The present invention requires trunk groups from PBXs to a switchoperated by the local exchange carrier (LEC) offering the features ofthe present invention. These trunk groups can be analog line-sideone-way or two-way trunk connections, Super Trunk or PRI ISDN. In mostcases the customer will be using PRI ISDN. In most cases (except foranalog lines), these trunks operate as two-way trunks; dynamicallocation on PRI; and ascending-descending on Super Trunk. These trunksmust tie into the local exchange carrier's provisioned switches in orderto trigger an event associated with the present invention. In caseswhere this PBX traffic is handled by private facilities as part of thecustomer's private network, e.g., ATM network, it is understood that alocal exchange carrier AIN provisioned central office is required to bethe first entry point in order to begin any event associated with thepresent invention. If not, the features of the present invention willnot be available to that station or trunk group.

[0122] Although the present invention may operate with 1AESS switches,manufactured by Lucent Technologies, Inc., in one embodiment, the 1AESSswitches are not configured to provide the features of the presentinvention. For example, when the 1AESS switch is not AIN provisioned.Thus, if the customer has centrex or PBX connection service out of oneof these switches, a trunk side connection is used to connect to anequipped switch, such as a DMS 100 or 5ESS digital switch 10, 12, wherethe triggers T are set. This is a trunk side connection to the nearestdigital office. In FIG. 1a, switch 10 is closest to PBX stations EB,while switch 12 is closest to centrex stations EA. In addition, localtraffic is sent back to the non-provisioned switch, e.g., the 1AESS,over the PSTN if the non-provisioned switch is the host switch for thecalled local telephone number.

[0123] In the case of independent local providers, the customer can usetwo devices to connect to the service. The customer can use either trunkgroup connections or a Link Extension. The Link Extension allowstelephone numbers to be provisioned from the serving central office inwhich the Link Extension terminates. This would force number changes onthese stations, however. The advantages to these two solutions should beweighed by the customer to determine which is most economical andeffective from a cost and user standpoint. In all cases it is to beexpected that the customer will bear the cost of any private lines runto connect to the service from tariffed products that are not part ofthe service.

[0124] With reference to FIG. 2, a common service logic for processingall calls placed by the customer is now described. More detail will beprovided about each of the major functions in later sections.

[0125] After the switch triggers and sends a query, the service controlpoint (SCP) processes the trigger and transfers to the common servicelogic for the subscribing customer. The primary purpose of the triggersis to transfer the queries to the specific customer's common servicelogic.

[0126] The common service logic initially determines the type of dialingplan. A simplified determination includes ascertaining the dialed accesscode (e.g., “7” or “9”) at step s100. For example, “7” can be associatedwith a manipulation dialing plan (MDP) and “9” can be associated with aspecial dialing plan (SDP). Although the following description is basedon these associations, any access code may be associated with eitherdialing plan. Moreover, although two specific dialing plans aredescribed, additional and alternate dialing plans are contemplated.

[0127] After the dialing plan is determined, the called number isdetermined. If the dialing plan was determined to be MDP, then at steps110 the called number is looked up in a table, based upon the dialednumber. If the dialing plan was determined to be SDP, then at step s112the called number is set to the 10 digit dialed number. After the callednumber is determined at either step s110 or s112, outgoing callscreening executes at step s120. Subsequently, at step s130 blockingtables are determined.

[0128] After the blocking tables are determined at step s130, it isdetermined whether automatic selection of facilities-remote tie line(ASF-RTL) is applicable at step s200. If ASF-RTL is not active, at steps202 normal PSTN processing proceeds. Subsequently, at step s204 PSTNspecific billing processing (i.e., usage billing suppression and usagebilling reduction) occurs. On the other hand, if ASF-RTL is active, atstep s208 the ASF-RTL processing occurs. Subsequently, ASF-RTL specificbilling (i.e., RBR) logic executes at step s210. Finally, at step s216SMDR data is generated and at step s218 the response is sent to theswitch.

Dialing Plan

[0129] According to an aspect of the present invention, a custom dialingplan is subscribed to by the customer, and the fees the customer paysfor the custom dialing plan are controlled by the UBS, UBR, and RBRbilling functions. The custom dialing plan allows customers to createtheir own abbreviated network dialing plan. In one embodiment, thecustom dialing plan consists of a combination of two dialing plans, the7+ manipulation dialing plan (MDP) and 9+ special dialing plan (SDP).SDP recognizes that 7 digit dialing plans no longer give large companiesthe flexibility that they need. SDP provides the flexibility whileoffering access to vertical features like routing, OCS, billing, andreporting.

[0130] MDP abbreviated numbers begin with the access code 7+, and permitfull manipulation. Full digit manipulation in MDP means the routingtelephone number can be either 7 digits (7D) or 10 digits (10D)regardless of what 7 digit or 10 digit telephone number was dialed.There need not be any digits from the dialed telephone number in therouting telephone number. Thus, MDP tables are provided that map thedialed number to the routing number. This degree of flexibility supports10 digit telephone numbers that may be either international or NorthAmerican from the customer's perspective.

[0131] Examples of 7 digit and 10 digit dialing in MDP follow: Customerdials 969-2300 and the call is routed to 213-562-2345 or customer dials1-213-969-2300 and the call is routed to 213-562-2345.

[0132] SDP uses 9+ as the access code, which is the access code to thepublic office dial plan (PODP). Thus, anything dialed after 9+ will behandled by the AIN. There is no digit manipulation much less full digitmanipulation. The dialed telephone number is the routing telephonenumber. Operator Services, carrier identification code (CIC) calls, andmiscellaneous N11 calls are simply passed through while accessmanagement is determined by privilege classes, which are discussedbelow.

[0133] According to the present invention, once the access code (9) or(7) has been dialed, the call will be routed out of the dialing party'slocal centrex or PBX. A second dial tone is thus required. It isunderstood that on any provisioned station or trunk group, dialing theleading digit (9) or (7) places the caller into the network of thepresent invention and therefore in the PSTN. If a private facility isthe routing choice, the call will hit the SSP in the triggering centraloffice prior to any routing.

[0134] The following rules apply in the custom dialing plan: On-netcalls are defined as being between a provisioned station and a numberlisted in the dialing plan table(s) and are always reached by dialingone of the access codes. Off-Net calls are defined as being between aprovisioned station and a number not listed in the dialing plan table(s)(i.e., Off-net telephone numbers in FIG. 1a). All calls On-Net andOff-Net are sent through outgoing call screening (discussed below) asappropriate. If the called number is not in one of the tables (i.e., thecall is an Off-net call), the call is not eligible for usage billingsuppression or usage billing reduction, although other features mayapply.

[0135] In order for certain per station features, such as outgoing callscreening and UBR, to operate on calls originating from a PBX telephoneon a station rather than trunk group basis, a caller ID function willneed to be activated on the ISDN PRI. Super Trunk and analog line sidetrunks either one way or two way will not have these features except ona trunk basis. It is noted that in the case of outgoing call screening,the caller ID function is provided by the PBX if the trunk is not PRIISDN with the caller ID function. Moreover, intracentrex calls andintraPBX calls will not trigger.

Outgoing Call Screening

[0136] According to an embodiment of the present invention, an outgoingcall screening feature is provided. Outgoing call screening is aprovisional optional feature implemented by service logic that executeswithin a SCP. That is, no centrex calls are screened in the switch. Inaddition, no PBX calls are screened in the switch or in the PBX. Withthe outgoing call screening feature, the subscriber assigns linerestrictions by assigning privilege classes to each station subscribingto the abbreviated dialing plan. Privilege classes can be defined by thecustomer to include the types of calls that each privilege class canmake. Privilege classes can be viewed as “how far” the user assigned tothe privilege class can call. In one embodiment, outgoing call screeningincludes the following privilege classes for On-net and Off-net callsshown in Table 1. It is noted that each privilege class has access toall destinations associated with the privilege class, in addition toaccess all destinations associated with the lower privilege classes.TABLE 1 Privilege Classes Privilege Class Privilege Title Description ofPrivilege Areas 8 Emergency calling and unrestricted Calls via anemergency private network 7 UNRESTRICTED All areas-worldwide access, 01,00, 0−, 0+ 6 Selected International Calls Off-net Calls to internationalcountries-same countries for entire privilege class 5 InternationalCalls On-net Calls to any On-net sites including international 4 AllNANP Numbers Off-net Calls to NANP Off-net numbers--same NANP for entireprivilege class 3 IntraLata - Off-net Calls to intraLATA numbers Off-net2 Selected International Calls On-net Calls to customer designatedinternational countries-same countries for entire privilege class 1Domestic On-net Calls within the MDP/SDP - On-net, 311, 411, 611, 911

[0137] According to an aspect of the present invention, blocking tablesare provided and associated with privilege classes. In one embodiment,separate blocking tables are provided to block international calls,domestic calls, and toll free number (hereinafter referred to as 1-800number) calls. An emergency function can also be provided, as can aninternational completion table, which lists selected internationaldestinations that can be called from, for example, privilege class 2stations in the exemplary configuration of Table 1. In an embodiment theblocking tables include up to 250 table entries.

[0138] Initially, a privilege class table is populated to define whichcall type codes (CTCs) are associated with each privilege class. A calltype code indicates the classification of a call, e.g., an On-netinternational call. Thus, a YES/NO value must be indicated for eachpossible call type code, for each privilege class (PC). Table 2 shows anexemplary Privilege Class Table. TABLE 2 PRIVILEGE CLASS TABLE Call TypeCode PC1 PC2 PC3 PC4 PC5 PC6 PC7 PC8 101 Y Y Y Y Y Y Y Y 109 Y Y Y Y Y YY Y . . . 131 N N N N Y Y Y Y 153 N Y Y Y Y Y Y Y etc.

[0139] In Table 2, call type code 101 represents On-net intraLATA calls.Call type code 109 represents On-net calls to 976 telephone numbers.Call type code 131 represents On-net interLATA domestic calls. Call typecode 153 represents On-net selected international calls. Additional calltype codes for each possible different call type can be defined and acorresponding value can be stored in the privilege class table for eachprivilege class.

[0140] Call type codes are predefined to readily identify allcombinations of call types. The call type codes derive from a nature ofnumber (NON) code. For example, in an embodiment of the invention, whenthe called party ID includes an NANP NON (e.g., 3), and the number planarea is 800, 888 or 877, the call type code may be determined by addingthe value of 41 to an initial value. The initial value may be 100 whenthe called number is an On-net call and the initial value may be 200when the called number is an Off-net call. Therefore, for an On-net 800number, the call type code would be 141 and for an Off-net 800 number,the call type code would be 241. Call type code values can be assignedto any type of calls that may be encountered. All of the examples buildon the NON and whether the called number is in On-net.

[0141] According to another embodiment of the present invention, aninternational completion table can also be provided. The internationalcompletion table stores a list of selected international numbers. Thus,privilege classes that permit completion to “selected” internationalnumbers (e.g., privilege class 2) can dial these selected internationalnumbers.

[0142] Subsequent to definition of the privilege class table, aprivilege class control table must be defined. Table 3 shows anexemplary privilege control table. The privilege class control tableindicates which (if any) blocking tables are associated with eachprivilege class. For example, privilege class 1 may be associated withan international blocking table, a domestic blocking table, and an 800number blocking table. Accordingly, any user assigned to privilege class1 will have all calls screened by each of the blocking tables. That is,if the dialed number is listed in any of the blocking tables, the callwill not go through and a message is played to the user informing thecaller of the blocking. A exemplary message is: “We're sorry but callsto that number cannot be made from your telephone. Please try again orcontact your supervisor for instructions.”

[0143] Exemplary blocking tables include a single column that stores allof the to be blocked. The entries can be either country codes, completeinternational telephone numbers, 10 digit NPANXX telephone numbers, 6digit NPANXXs, 3 digit NPAs, or complete 10 digit 800 telephone numbers.TABLE 3 PRIVILEGE CONTROL TABLE PRIVILEGE INTERNATIONAL DOMESTIC 800CLASS EMERGENCY BLOCKING BLOCKING BLOCKING 1 N Y Y Y 2 N Y Y Y 3 N Y Y Y4 N Y Y Y 5 N Y Y Y 6 N Y Y Y 7 N Y Y Y 8 Y N N N

[0144] An emergency calling feature can also be facilitated by theprivilege control table. According to this embodiment, when emergencycalling is in effect, all calls are blocked when a privilege class has aNO entry in the “Emergency” column (privilege classes 1 -7 in Tables 1and 3). If the entry for a privilege class is YES in the emergencycolumn (privilege class 8 in Tables 1 and 3) then calls are not blockedwhen emergency calling is in effect.

[0145] The emergency calling feature operates to affect calling partieswho are assigned a privilege class that normally allows calls tocomplete. When emergency calling is in effect, the affected callingparties are prevented from completing the normally allowed calls so thatprivate trunk groups are available during emergencies for callingparties with the correct privilege class. Use of the emergency callingfeature blocks most calling party numbers from completing calls. Thecalling party numbers not blocked will have their calls routed over theprivate trunk groups.

[0146] Emergency calling is valuable if the customer has a small privatenetwork to ensure critical traffic can get through when publicemergencies are overloading the PSTN. In this case, the private trunkgroups constituting this small private network are the first choiceroute in all switches for all triggering telephone numbers, withoverflow set up to other private trunk groups, or to carriers, or to thePSTN.

[0147] In an embodiment, telephone numbers, such as 7 or 9 +1+800NXX-XXXX, are available to all privilege classes and are typicallynot blocked unless identified in the 800 blocking table. On the otherhand, all privilege classes are restricted from calling NPA-900 andNNX-976 numbers. Customer assigned NPA-900 and NNX-976 numbers that needto be accessed on a system wide basis are contained in the MDP table asterminating only telephone numbers or tables supporting SDP.Consequently, these customer assigned NPA-900 and NNX-976 numbers can beaccessed. In CTC (outgoing call screening-Call Type Code) terms, 900/976On-net is YES for all privilege classes (in the shown example), and900/976 Off-net is NO for all privilege classes (in another exemplaryembodiment).

[0148] The abbreviated dialing plan service subscriber may assign aprivilege class to any centrex station. For PBX, the privilege class isassigned to the trunk group, unless the trunk group is primary rate ISDN(PRI) with a caller ID function provisioned per station.

[0149] With reference to FIG. 3, an exemplary logic flow to implementoutgoing call screening is described. The logic is called from step s120in FIG. 2. Initially, at step s1000 it is determined whether thecustomer subscribed to the outgoing call screening feature. If thecustomer did not subscribe to outgoing call screening, at step s1002 thelogic returns to FIG. 2 to proceed. If the customer did subscribe tooutgoing call screening, at step s1004 the calling party number'sprivilege class and type is determined. That is, the values are lookedup in outgoing call screening tables 1006.

[0150] In one embodiment, the outgoing call screening tables include asingle index table and multiple outgoing call screening tables. Theindex table is used to determine which outgoing call screening tablestores the 10 digit calling party number based on some portion of theNPANXX of the calling party number. Each outgoing call screening tablestores a portion of the 10 digit calling party numbers, depending on thecapacity of the table structure and logical groups that may be madebased on the first digit, first two digits, first three digits, firstfour digits, first five digits, and first six digits, such that thecapacity is not exceeded.

[0151] The correct outgoing call screening table is identified bysearching the index table. The search uses the NPANXX of the callingparty number to find the name of the outgoing call screening table. Ifthe name is found, the identified outgoing call screening table issearched to find the corresponding privilege class. Otherwise, the NPANXof the calling party number is used. If the name is found, theidentified outgoing call screening table is searched to find thecorresponding privilege class. Otherwise, the NPAN of the calling partynumber is used. If the name is found, the identified outgoing callscreening table is searched to find the corresponding privilege class.Otherwise, the NPA of the calling party number is used. If the name isfound, the identified outgoing call screening table is searched to findthe corresponding privilege class. Otherwise, the NP of the callingparty number is used. If the name is found, the identified outgoing callscreening table is searched to find the corresponding privilege class.

[0152] If the name is still not found, the N of the calling party numberis used. If the name is found, the identified outgoing call screeningtable is searched to find the corresponding privilege class. Otherwise,the called party ID is set to the 10 digit telephone number of astandard vacant number announcement, and the standard vacantannouncement is played. Subsequently, the logic proceeds to step s218 inFIG. 2. In this last case, a provisioning error occurred. That is, thecalling party number was not added to the correct outgoing callscreening table with the privilege class.

[0153] In each of the above cases, if no corresponding privilege classis found in the identified table, a default privilege class is assigned.In addition, the outgoing call screening table identifies each callingparty number as either PBX, centrex, or terminating only. Thus, a calltype variable (not to be confused with the call type code) associatedwith the calling party number is set in accordance with thecorresponding call type determined from the outgoing call screeningtable.

[0154] After the privilege class and type are identified, at step s1008the call type code is determined. The determination is described indetail later. After the call type code is determined, it is determinedwhether the privilege class includes the call type code, at step s1010.More specifically, the privilege class determined at step s1004 and thecall type code determined at step s1008 are used to perform a lookup inthe privilege class table. If the lookup returns a YES, at step s1012the call continues processing according to the logic of FIG. 2, i.e.,the logic flows to step s130. If the lookup returns a NO, at step s1014the call is blocked and the appropriate announcement is played.

[0155] As indicated in FIG. 2, the blocking tables are determined nextat step s130. The blocking tables feature requires an entry in theprivilege control table, blocking tables listing the numbers to whichcalls are to be blocked, and service logic. The service logic checks, ona real time basis, the dialed number to see if the dialed number shouldbe blocked.

[0156] More specifically, when the call type code corresponds to one ofthe blocking table types (e.g., international, domestic, and 800), thecommon service logic looks up a value in the privilege control table.Next, it is determined if the called number corresponds to one of theblocking table types. That is, it is determined whether the callednumber is an 800 number, domestic number, or international number. Ifthe called number corresponds to one of the blocking tables, it is thendetermined from the privilege control table whether a YES valuecorresponds to the privilege class. If a YES value is found, the callednumber is validated against the appropriate blocking table. Otherwise,if a NO value is found in the privilege control table, validation of thecalled number is bypassed.

[0157] In one embodiment, validation of international numbers occurs byfirst checking the entire dialed number to see if it is in theinternational blocking table. If the dialed number is in theinternational blocking table, an announcement is played informing thecaller that the call is blocked. If the dialed number is not found, thefirst three digits of the dialed number are searched in theinternational blocking table. If the first three digits of the dialednumber are in the international blocking table, an announcement isplayed informing the caller that the call is blocked. If the first threedigits of the dialed number are not found, the first two digits of thedialed number are searched for in the international blocking table. Ifthe first two digits of the dialed number are in the internationalblocking table, an announcement is played informing the caller that thecall is blocked. If the first two digits of the dialed number are notfound, the first digit of the dialed number is searched for in theinternational blocking table. If the first digit is not found, theinternational blocking table logic is complete and the call can proceed.If the first digit is found, an announcement is played informing thecaller that the call is blocked.

[0158] In one embodiment, validation of domestic numbers occurs by firstchecking the NPANXXXXXX to see if it is in the domestic blocking table.If the NPANXXXXXX is in the domestic blocking table, an announcement isplayed informing the caller that the call is blocked. If the NPANXXXXXXis not found, the NPANXX of the dialed number is searched for in thedomestic blocking table. If the NPANXX of the dialed number is in thedomestic blocking table, an announcement is played informing the callerthat the call is blocked. If the NPANXX of the dialed number is notfound, the NPA of the dialed number is searched for in the domesticblocking table. If the NPA of the dialed number is in the domesticblocking table, an announcement is played informing the caller that thecall is blocked. If the NPA of the dialed number is not found, thedomestic blocking table logic is complete and the call can proceed. Ifthe NPA is found, an announcement is played informing the caller thatthe call is blocked.

[0159] In one embodiment, validation of 800 numbers occurs by firstchecking the 800NXXXXXX to see if it is in the 800 blocking table. Ifthe 800NXXXXXX is in the 800 blocking table, an announcement is playedinforming the caller that the call is blocked. If the 800NXXXXXX is notfound, the 800NXX of the dialed number is searched for in the 800blocking table. If the 800NXX of the dialed number is in the 800blocking table, an announcement is played informing the caller that thecall is blocked. If the 800NXX of the dialed number is not found, the800 blocking table logic is complete and the call can proceed.

ASF-RTL

[0160]FIG. 4 illustrates isolated portions of an exemplarytelecommunications network in association with the present invention,for implementing the ASF-RTL feature in particular. The network includesoriginating service switching points (SSPs) 2002, 2004 and 2006. SSP2004 is shown to have a corresponding centrex system 2010, includingcalling party telephone 2011, and a PBX 2012. The other SSPs 2002 and2006 likewise have associated centrex systems and/or PBXs, although theyare not described further. Also shown in FIG. 4 is an ASF-RTL host SSP2014 and an ETN node 2016. The ETN node 2016 is connected to aterminating SSP 2018 in a private network separate from that in whichSSPs 2002, 2004 and 2006 are located, as well as an internationalgateway 2026. The telecommunications network also includes a localsignal transfer point (STP) 2020, a regional STP 2022 and a servicecontrol point (SCP) 2024.

[0161] The present invention operates in an AIN environment. All of theswitches, including SSPs 2002, 2004, 2006 and 2018 and the ASF-RTL hostSSP 2014 are therefore AIN compatible switches, such as the 5ESS, or theDMS-100 switches. All of the switches are configured to receive andtransmit either 7 digit or 10 digit SS7 signaled messages. The operationof ASF-RTL, is dependent on the transmission of 10 digit SS7 signalingsuch that the true called party number (CDN), originally dialed by thecalling party, can be ultimately transmitted. The ETN node is likewise a5ESS or DMS-100 switch in an embodiment of the invention. In analternative embodiment, an ATM switch is substituted for the ETN node.The ATM switch performs a function similar to the ETN node with respectto the present invention, and may be a MainStreetXpress 36170Multiservices Switch or 670 RSP, both manufactured by Newbridge NetworksCorporation; a GX 550 Smart Core ATM Switch, manufactured by LucentTechnologies Inc.; or a Passport 15000 Multiservice Switch, manufacturedby Nortel Networks Corporation.

[0162] For purposes of description, SSP 2004 is an exemplary originatingcentral office (CO) for the calling party telephone 2011 and SSP 2018 isthe terminating central office for the called party telephone 2031. SSP2014 is the host central office for the ASF-RTL system, and has a uniquetelephone number accessible by the originating central office 2004.Generally, the SSPs supporting the network shown in FIG. 4 launchqueries to the SCP 2024, via STPs 2020 and 2022 over SS7 signal links(shown as small dashed lines), which translates abbreviated dialednumbers into numbers understood by the PSTN. These translations arehighly centralized in that they only occur at the SCP 2024, but becausethe responses are directed to the various querying switches, thepractical effect is highly decentralized.

[0163] Also shown in FIG. 4 are the various connections among the systemelements. All private trunks, shown connecting the ETN node 2016 withthe ASF-RTL host SSP 2014 and with the international gateway 2026, aremulti-frequency (MF) trunks (shown as large dashed lines). Therefore,they do not send the calling party number (CPN). However, calls goingover the PSTN have a high degree of likelihood of being carriedend-to-end via SS7 trunks (shown by as bold lines) and SS7 links (shownas small dashed lines), and hence, the calls deliver the calling partynumber if the called number has associated services requiring this data,such as caller ID or incoming call screening.

[0164] The present invention requires trunk groups from the centrex andPBX sites to a switch (shown as thin solid lines), operated by the localexchange carrier (LEC). These trunk groups can be analog line-sideone-way or two-way trunk connections, Super Trunk or PRI ISDN. Thesetrunks tie into the LEC's provisioned switches in order to trigger anevent associated with the present invention. In cases where this trafficis handled by private facilities as part of the customer's privatenetwork, it is understood that a local exchange carrier AIN provisionedcentral office is required to be the first entry point in order to beginany event associated with the present invention.

[0165] According to the present invention, three types of routing areprovided: centralized route selection (CRS), centralized carrierselection (CCS) and ASF-RTL. These features are based on the protocol ofAIN R0.1, and permit the SCP to send up to three private truck groupsand three carrier identification codes (CICs) in one response to an SSPquery. Any private trunk groups returned by the SCP to a switch takeprecedence over any CICs returned at the same time. These features allowintegration of a private network as the customer may desire for reasonseither based on economics or service criticality.

[0166] Centralized route selection integrates private facilities intothe service of the present invention. Centralized route selection allowsup to three route selections per destination. Routing may also be basedon time of day/day of week (tod/dow) decisions, type of call (CTC),number plan area (NPA), date, and percentage allocation.

[0167] Centralized carrier selection allows up to three carrierselections per line. It provides two alternate carrier selections incase of failure on the primary carrier's network. Centralized carrierselection also allows the customer to use multiple carriers on a tow/dowbasis, as well as CTC, NPA, date, and percentage allocation routing.

[0168] ASF-RTL is the third possible type of call routing, whichcombines trunk group and carrier selections. It is available regardlessof whether the call is dialed using MDP or SDP. Generally, underASF-RTL, call traffic from an originating SSP is directed to acentralized hub node, referred to as an ASF-RTL host SSP, via the PSTN.The ASF-RTL host SSP must have an assigned telephone number for routingpurposes. From the ASF-RTL host SSP, the calls are routed to eitherprivate facilities (i.e., private trunk groups), carrier facilities(i.e., an IXC POP), or an international gateway. The call destinationdetermines which calls will be routed to the ASF-RTL host SSP, describedbelow. Generally, calls identified as routing to the ASF-RTL host SSPare all international calls and calls directed to major PBX or centrexsites, inside or outside the LATA of the originating number.

[0169] Routing features within the ASF-RTL system allow the customermaximum flexibility in choosing private facilities or the PSTN, usingexisting lines, or allowing the customer to establish new routing basedon cost effectiveness between locations. In the event that the trafficcrosses LATA boundaries, the call must be able to be completed by aninterexchange carrier. In the event of congestion or failure on theASF-RTL facilities, the call will receive an all circuits busy signal.Carrier selection can also be used to route traffic away from theASF-RTL telephone number in an overflow situation (if the routing can beperformed over the PSTN).

[0170] Referring to FIG. 4, ASF-RTL recognizes that most small locationshave no need for a direct, private trunk group from that location to anETN node (or ATM switch). However, even without direct private trunkgroups, the traffic from the small locations tends to go the same ETNnode. For example, without the ASF-RTL service, calls over the PSFNpassing through originating switches 2002, 2004 and 2006 would naturallypass through a common switch, such as ASF-RTL host SSP 2014, to reachthe ETN node 2016. If the call traffic were placed on direct privatetrunk groups, the direct private trunk groups would eventually runparallel to each other, beginning at the common switch.

[0171] Therefore, ASF-RTL permits the call traffic from most smalllocations to “ride” the PSTN until the point where the parallel trunkgroups would likely begin (i.e., host SSP 2014). At that point, aprivate trunk group, i.e., “tie line,” connects the ASF-RTL host SSP2014 to the ETN node 2016, indicated by the large dashed line betweenSSP 2014 and ETN node 2016 in FIG. 4. Because the calls from multiplelocations are aggregated at the SSP 2014, there is sufficient traffic tojustify the expense of incorporating a private trunk group beyond thatpoint. Meanwhile, the SS7 trunk, shown by the bold line between SSP 2014and ETN node 2016, is still available to handle any overflow traffic, asdetermined by the ASF-RTL service logic, discussed below.

[0172] From an AIN perspective, the call traffic subject to ASF-RTLrouting is identified when the originating trigger occurs at SSP 2002,2004, or 2006. The trigger information is ultimately received by theintermediate terminating SSP 2014, which hosts the start of the privatetrunk group. The traffic is placed on the aggregating private trunkgroup by means of a route index, for example.

[0173] Therefore, when ASF-RTL is active, the SCP 2024 receives a queryfrom an originating switch (e.g., SSP 2004), which includes the true“called party number”(e.g., the telephone number of the destinationtelephone 2031). The SCP 2024 returns the called party number to theoriginating switch as a redirecting number (RDN), along with a number ofthe switch hosting the private ASF-RTL facility group to be entered(e.g., SSP 2014) as the new called number. Therefore, the parametersappearing in the SS7 message at the SSP 2014, hosting the ASF-RTLtelephone number, include a calling party ID (i.e., the 10 digitoriginating telephone number), the called party ID (i.e., the 10 digitASF-RTL telephone number) and the original called party ID (i.e., thenumber originally dialed and set as the redirecting number in theoriginating SSP).

[0174] The ASF-RTL host SSP 2014 will then trigger an SCP, which forpurposes of discussion is SCP 2024, although any compatible SCP in theservicing area is suitable.

[0175] The SCP 2024 executes service logic, which mimics outgoing callscreening's service logic, to obtain call type codes and routing optionsto determine which facility grouping to use. The logic is described indetail below. Normally, a facility grouping will have one to three routeindices for private trunk groups, but it may also have one or more CICsfor overflow purposes. The service logic discerns interLATA fromintraLATA connections, so that the overflow CICs can be IXCs forinterLATA and private networks for intraLATA, for example. Finally, theSCP 2024 responds to the query with the redirecting number received inthe query from the originating SSP 2004 as the CDN. As in centralizedroute selection and centralized carrier selection, ASF-RTL may includerouting according to type of call, NPA, specific dates, day of week,time of day, and percentage allocation.

[0176]FIG. 5 shows a flowchart of the ASF-RTL service logic executed bythe SCP 2024 to route call traffic according to the present invention.The ASF-RTL service logic is initiated when the SCP 2024 executes steps208 of FIG. 2. Generally, automatic selection of facilities (ASF) is atechnical vertical feature for providing centralized route selection,centralized carrier selection, or ASF-RTL, which apply to call routing.ASF bundles the features so that routing can efficiently be determinedon a per call basis. ASF-RTL is exclusive of centralized route selectionand centralized carrier selection, so the steps of FIG. 5 are executedonce the decision has been made to follow ASF-RTL as opposed to thealternative call routing services: centralized route selection andcentralized carrier selection.

[0177] ASF is dependent on the originating location. When a call isplaced, the common service logic determines whether ASF exists at theoriginating telephone number, and if so, whether ASF-RTL is availablefor routing. If ASF-RTL is available, the ASF-RTL service logic of FIG.5 is initiated by the SSP hosting the ASF-RTL telephone number (i.e.,ASF-RTL host SSP 2014). The ASF-RTL host SSP 2014 triggers at step s2110and sends a query to the SCP 2024. Each ASF-RTL telephone number has itsown trigger, although the ASF-RTL service logic is common to all ASF-RTLtelephone numbers for a specific customer.

[0178] At step s2112 a security check is performed by the SCP 2024.Generally, a security check is not necessary because an ASF-RTLtelephone number does not send a query unless appropriately triggered.However, it is possible to dial the ASF-RTL telephone number of the hostswitch directly (i.e., from outside the call routing service). Forexample, telemarketers may dial the ASF-RTL telephone number even thoughit is not a public number. The security check therefore assures that acall entering the ASF-RTL host switch 2014 is initiated from anauthorized workstation. Because the telephone number will trigger forall calls to it, the security is based on the calling party number.

[0179] In an embodiment of the invention, the security check begins withthe ASF-RTL service logic validating the calling party. If the callingparty is not authenticated (s2114: NO), an announcement notifying thecaller that the call cannot be completed is set and a return response issent to the SSP 2014. The SSP 2014 plays an announcement (e.g., theannouncement of step s2124), stating that the attempted call is notauthorized.

[0180] Once the called party is authenticated (s2114: YES), the ASF-RTLservice logic obtains the nature of the number (NON) indicated by thecalled party ID. For example, NON may be a 4 to indicate aninternational call, a 3 to indicate a North American Numbering Plan(NANP) call, and a 1 for a intra-customer network call.

[0181] After authentication, the ASF-RTL service logic determines thecall type code (CTC) at step s2116. Determining the call type code isaccomplished by separate service logic, which is unique to eachcustomer. However, the same call type code service logic is used totrigger all telephone numbers of the customer, including the ASF-RTLtelephone numbers, in order to ensure consistent call type codedeterminations for the same call regardless of the features includedwith the call. The call type code service logic ultimately returns thecall type code to the ASF-RTL service logic.

[0182] The ASF-RTL system processes calls according to the call typecode. For example, in one embodiment of the invention, call type codesindicating intraLATA centrex to centrex, centrex to PBX, and PBX to PBXcalls, calls to 800 numbers, On-net 900/976 calls, Off-net local calls,Off-net CIC calls, and operator assist calls are allowed to completeover the PSTN. Consequently these types of calls are not routed to theASF-RTL telephone number. Call type codes indicating calls to Off-net900/976 telephone numbers are blocked, requiring no route selectionprocessing. The remaining call type codes, including call type codesindicating international calls and interLATA domestic calls, are routedto the ASF-RTL telephone number. Examples of call type codes subject toASF-RTL processing, according to the present example, include 101(On-net, intaLATA), 151 (On-net, international), 252 (Off-netinternational), 131 (On-net, intrastate, interLATA) and 231 (Off-net,intrastate, interLATA).

[0183] Once the call type code has been determined, the ASF-RTL servicelogic prepares data for the routing decision options at step s2118 ofFIG. 5. Each ASF-RTL telephone number has its own associated routingdecision logic. Therefore, step s2118 ensures that the necessaryinternal call variables exist and are properly passed to the ASF-RTLrouting service logic at step s2120. The necessary call variablesinclude an originating-location variable, which is set by the servicelogic to equal the ASF-RTL telephone number. Another call variablecontains the results of the ASF-RTL routing decision. The ASF-resultscall variable has a default value of zero, which indicates that theresponse to the SSP is to include only the called party ID (which hasbeen set back to the called number) as the PSTN default route.

[0184] Once the variables have been set, the ASF-RTL service logicpasses control to the ASF-RTL routing service logic at step s2120. Steps2120 is the most flexible aspect of ASF-RTL and does not followprescribed paths. The purpose of the ASF-RTL routing service logic is toreturn an outcome to the ASF-RTL service logic for determining the routeof a call. The ASF-RTL routing service logic is a series of logicdecision nodes and branches provisioned on a custom basis for eachlocation, resulting in multiple branch ends, as indicated in FIG. 6.Each branch end may alone be an outcome, or branch ends may be tiedtogether to share an outcome. If any of the possible results reach areturn node, returning to the ASF-RTL service logic, without specifyingan outcome, a default occurs, resulting in blocking of the call.

[0185] Each outcome determined by the ASF-RTL routing service logic mustbe matched by a row in an ASF-RTL table stored at the SCP 2024, whichincludes at least an entry for the originating-location call variableand a destination-location call variable. In an embodiment, the ASF-RTLtable also includes trunk and carrier options, such as a primary trunk,an alternate trunk, and a second alternate trunk, and a primary carrier,an alternate carrier and a second alternate carrier. An exemplaryASF-RTL table is shown in Table 4, below.

[0186] Execution of the ASF-RTL routing service logic of step s2120results in an outcome corresponding to a destination-location callvariable, which identifies the route for reaching the destination of thecall (i.e., the called number). The default value of thedestination-location call variable is none, or zero. Any value otherthan default is assigned at a branch end of the decision optionsdetermining the outcome, as shown in FIG. 6. If the default value is notchanged by the execution of the ASF-RTL routing service logic, the callwill be blocked.

[0187] Determination of the destination-location call variable is basedon the call type code of the called number, indicated at step s2310 ofFIG. 6. Alternatively, determination of the destination-location callvariable can be based on time of day/day of week, or any other factor,instead of the call type code. The call type code has previously beendetermined at step s2116 of FIG. 5. The ensuing processing results inone of three possible outcomes, in the depicted embodiment of theinvention, each providing a distinct destination-location call variable.For example, if the call type code indicates an On-net intaLATA call(e.g., 101), the process proceeds to blocks s2312, which results insetting the destination-location variable at s2318 to Outcome 1. If thecall type code indicates a selected On-net international call, such as153 or selected Off-net international call (e.g., 253), the processproceeds to blocks s2320 and s2322, respectively, which result insetting the destination-location call variable at s2328 to Outcome 2.Lastly, if the call type code indicates an On-net interLATA domesticcall, such as 131, or an Off-net interLATA domestic call, such as 231,the process proceeds to blocks s2340 and s2342, respectively, whichresult in setting the destination-location call variable at s2348 toOutcome 3. If the call type code is some other call type not subject toASF-RTL processing, at step s2350 no outcome is determined and the callis default routed to the PSTN. All of the outcomes, including defaultindications, are returned at s2352 of FIG. 6, returning the process tostep s2120 of FIG. 5.

[0188] At step s2122 of FIG. 5, the ASF-RTL route is determined usingthe call variables previously generated. This step begins with theASF-RTL service logic checking the outcome returned from step s2120,identifying the value of the destination-location call variable anddetermining whether the value is still the default value. If there is avalue other than the default value, a look-up in the ASF-RTL table isperformed. If the destination-location call variable is the defaultvalue, then the look-up in the ASF-RTL table is skipped.

[0189] The ASF-RTL table contains rows of data corresponding to thedestination-location and originating-location call variables. Table 4 isan exemplary ASF-RTL table, showing possible table entries correspondingto the outcomes produced by the ASF-RTL routing service logic, accordingto one embodiment of the invention: TABLE 4 Originating DestinationPrim. Alt. 2nd Alt. Pri. Alt. 2nd Alt. Location Location Trunk TrunkTrunk Carrier Carrier Carrier (925) TG 0000- 789-5678 0174 (925) TGCIC0000- 0000- 0222 789-5678 0174 0174 (925) TGCIC2 0000- 0195- 0000- 02220333 0388 789-5678 0174 2000 0180

[0190] The rows of Table 4 include the designated trunks and carriersassociated with the destination-location call variable (and associatedoriginating-location call variable). The various values are simplyexamples of predetermined identification numbers corresponding todesired trunk group and carriers. A successful look-up of the ASF-RTLtable produces a trunk and/or a carrier for call routing. If a row ofdata is not retrieved, a critical error has occurred that will result inan appropriate announcement being played by the SSP.

[0191] Because the ASF-RTL system supports up to three private trunkgroups, the table may include an alternate trunk group and a secondalternate trunk group, in addition to the primary trunk group. Thealternate trunk group entry can appear only when a primary trunk groupis provided; likewise, the second alternate trunk group entry can appearonly when an alternate trunk group is provided. Similarly, because theASF-RTL system supports up to three carriers, the table may include analternate carrier and a second alternate carrier group, in addition tothe primary carrier, to handle call overflow situations. The alternatecarrier entry can appear only when a primary carrier is provided;likewise, the second alternate carrier entry can appear only when analternate carrier is provided. The three carriers are included foroverflow situations in which the three private trunk groups specifiedare busy.

[0192] The three rows of data in Table 4 correspond to Outcome 1,Outcome 2 and Outcome 3 of FIG. 4, respectively. Outcome 1 is simply aprivate trunk group (e.g., the primary trunk), which is consistent withan On-net intraLATA call. In particular, On-net intraLATA calls stay onprivate trunk groups because an outside carrier would not have therequisite logic to handle such calls. Therefore, when a call type codevalue indicates an On-net intraLATA call, the destination-location entryin the ASF-RTL table does not have any associated carrier entries.Outcome 2 is a trunk group/CIC combination, including primary andalternate trunk group identification numbers and primary and secondarycarrier values. Outcome 3 is a trunk group/CIC combination, includingprimary, alternate and second alternate trunk group identificationnumbers and primary, alternate and second alternate carrier values.

[0193] The ASF-RTL service logic checks the values of the trunk groupand carrier call variables to ensure that the row of the ASF-RTL tableis valid. If either the primary trunk group entry or the primary carrierentry has a value, the row is deemed valid and the ASF-results callvariable is set. If both the primary trunk group entry and the primarycarrier entry are blank, the row is deemed invalid and an appropriateannouncement is instructed to be played by the ASF-RTL host SSP.

[0194] The ASF-results call variable is set to a predetermined numberindicating the number of trunk and carrier entries existing in the rowcorresponding to the destination-location table entry. Table 5 is anexemplary ASF-results table, showing possible entries according to oneembodiment of the invention: TABLE 5 ASF-Results: 1 2 3 10 11 12 13 2021 22 23 30 31 32 33 Pri. Trunk x x x x x x x x x x x x Alt. Trunk x x xx x x x x 2d Alt. Trunk x x x x Pri. Carrier x x x x x x x x x x x xAlt. Carrier x x x x x x x x 2d Alt. Carrier x x x x

[0195] In the embodiment of the invention depicted by Table 5, theASF-results values are determined according to the number of routingparameters provided for a particular option in the ASF-RTL table (Table4). The ASF-results values, indicated in the first row of Table 5, aresignificant in that the ASF-RTL host SSP 2014 will use only thoserouting parameters indicated to be available by the ASF-results values.For instance, an ASF-results variable having a single digit value of 1,2 or 3, or a double digit value with a 1, 2 or 3 as the second digit(i.e., 11, 12, 13, 21, etc.), indicates a primary trunk group isavailable. An ASF-results variable having a single digit value of 2 or3, or a double digit value with a 2 or 3 as the second digit, indicatesthat an alternate trunk group is available. An ASF-results variablehaving a single digit value of 3, or a double digit value with a 3 asthe second digit, indicates that a second alternate trunk group isavailable.

[0196] Similarly, with respect to carriers, an ASF-results variablehaving a double digit value with a 1, 2 or 3 as the first digit (e.g.,10, 20, etc.), indicates that a primary carrier is available. AnASF-results variable having a double digit value with a 2 or 3 as thefirst digit, indicates that an alternate carrier is available. AnASF-results variable having a double digit value with a 3 as the firstdigit, indicates that a second alternate carrier is available. Usingthis coding, the ASF-results variable for Outcome 1, Outcome 2 andOutcome 3 of Table 4 would be 1, 12 and 33, respectively.

[0197] The ASF-RTL service logic returns the ASF-results variable to thecommon service logic, along with the parameters of the trunks andcarriers for call routing, at step s2124 of FIG. 5. The SCP 2024 directsthe call through instructions to the ASF-RTL host SSP 2014 using thesevalues, along with the calling party ID.

[0198] An exemplary call routing using the ASF-RTL system is shown inthe call flow diagram of FIG. 7, according to an embodiment of theinvention. The calling party 2011, having the telephone number925-834-1234, initiates a telephone call at step 2402 by dialing thedestination telephone 2031, having the telephone number 415-921-5678.Originating SSP 2004 is triggered by the telephone call, and queries SCP2024 at step 2404 for call processing instructions. The query includesthe initiating telephone number 925-834-1234 as the calling party numberand the dialed number 415-921-5678 as the called number. The query alsoincludes the nature of the number called, which for purposes of theexample is a 3 (i.e., call within the NANP).

[0199] The SCP 2024 proceeds to look-up the called number in the MDPtable to determine the called party ID, which in this case is also415-921-5678. If the dialed number was 7 digit number, the SCP 2024would have determined the 10 digit value according to the MDP table. TheSCP 2024 then looks up the called party ID in an ASF table to identify acorresponding ASF-RTL telephone number, if any, which in this example is925-789-3412. The ASF-RTL telephone number is the telephone number ofthe ASF-RTL host SSP 2024. Because the called number is in the ASF-RTLtable, the SCP 2024 changes the called party ID to the ASF-RTL telephonenumber, 925-789-3412, reclassifies the original called number as theoriginal called number (ON), or the redirecting number (RDN), assigns anoriginal called party ID equal to the ON, and returns the called partyID, the original called party ID and the calling party ID to SSP 2004 atstep 2406.

[0200] Based on the instructions returned from the SCP 2024, theoriginating SSP 2004 determines whether the new called party ID (i.e.,the ASF-RTL telephone number) is an intraLATA or an interLATA call. Itgenerates the initial address message (IAM) for SS7 signaling and routesthe call via the trunk identified in the IAM to the second switch, SSP2014, which is the host switch for the ASF-RTL private trunk group, atstep 2408. The SSP 2004 likewise forwards to the ASF-RTL host SSP 2014the call routing instructions, including the called party ID, theoriginal called party ID, the calling party ID, the called number NONand the original called number NON. The SSP 2014 responds with anaddress completion message to confirm receipt of call data at step 2410.

[0201] The ASF-RTL host SSP 2014 then queries the SCP 2024 at step 2412for instructions, including in the query the call routing instructionsreceived from the SSP 2004. In a collected address variable, the SCP2024 is informed that the querying switch is the host switch for theASF-RTL private trunk group. The SCP 2024 then executes the ASF-RTLservice logic, previously described in FIGS. 6 and 7. For example, theSCP 2024 performs a security check, discussed with reference to FIG. 5above, prior to instructing the switch to forward the call over thetie-line. The SCP 2024 then determines the trunk groups and carriersover which the call may be forwarded. The called party ID is changedback to the original called number, which is sent, along with thecalling party ID, the trunk group ID and the called number NON back tothe ASF-RTL host SSP 2014 at step 2414.

[0202] The ASF-RTL host SSP 2014 determines whether the primary trunkgroup is busy, or otherwise unavailable. If so, the SSP 2014 may routethe call according to any alternate trunk group or carrier parametersprovided by the SCP 2024. The SSP 2014 determines the route index andforwards the called party ID over the determined trunk group (andcarrier, if the PSTN is used) to the ETN node 2016 at step 2416, via MFtrunk signaling. The ETN node 2016 forwards the called party ID to theterminating switch, SSP 2018, at step 2418, also via MF trunk signaling.The SSP 2018 terminates the call to the called party telephone 2031 atstep 2420. When the call is answered, an answering indication isreturned to the ASF-RTL host SSP 2014 at step 2426, via the SSP 2018 atstep 2422 and the ETN node at step 2424. The final “talk path”connection between the calling party 925-8734-1234 and the called party415-921-5678 incorporates call routing through the originating SSP 2004,the ASF-RTL host SSP 2014, the ETN node 2016 and the terminating SSP2018.

[0203]FIG. 8 is a call flow diagram depicting an exemplary call routingof an international call using the ASF-RTL system. Steps 2402 through2414 in FIG. 8 are identical to those steps of FIG. 7, except that thecalled number NON and the original called number NON are equal to 4,indicating an international call. Moreover, at steps 2406-2414, thecalled number NON is temporarily set to 3, indicating that the ASF-RTLtelephone number is an NANP number. After the call is routed to theASF-RTL host SSP 2014 at step 2414, the SSP 2014 routes the call to aninternational gateway 2026, as opposed to the ETN node 2016. Theinternational gateway 2026 routes the call, based on the called party IDto the appropriate international destination.

Usage Billing Treatment

[0204] As briefly discussed above, usage billing treatment includes twofunctions: usage billing suppression and usage billing reduction. Withthese two options, the customer is able to choose the appropriate usagetreatment based on their usage patterns and overall network trafficconfigurations.

[0205] Usage billing treatment from centrex and PBX stations can beapplied on centrex to centrex calls, centrex to PBX calls, PBX to PBXcalls, and PBX to centrex calls. The billing treatment provides thecustomer's account team with flexibility to use these treatment featuresin order to discount usage pricing depending on current and projectedusage trends. The highest grade of discount is usage billingsuppression, in which the billing record is discarded and the usage ischarged as a usage billing suppression feature charge or flat rateprice.

[0206] Usage billing reduction allows usage to be rated at a discountprice and allows flexibility in charges based on amount of usage,network configurations, competitive response, and which fullyparticipating classes of service originate the call and which receivethe call. Usage billing reduction pricing is set by contract usagepricing but is supported by billing systems.

[0207] At all times these usage billing treatment classifications forcentrex and PBX are only available on like classes of service if alloriginating and destination stations, trunks or DID numbers areprovisioned with the same treatment classification. In the case of likeclasses of services or unlike classes of services having differenttreatment classification, the lowest grade of discount will apply, i.e.,the highest usage price. In any case, all pricing paradigms must besupported by billing systems.

[0208] Usage billing suppression (UBS) is an option for the triggeringcentrex stations in a subscribing customer group. In order to beeligible for usage billing suppression, the originating and receivingstation lines must be fully participating centrex stations and the callmust be a voice only call and an intraLATA call.

[0209] Usage billing reduction (UBR) is an option for the triggering PBXstations/trunk groups (DID banks) in a subscribing customer group. Usagebilling reduction is available between all provisioned stations whetherthey are PBX or centrex as long as they are provisioned with UBR or UBS,respectively, and the call is intraLATA. In order to be eligible for UBRthe calling station has to be a fully participating station/trunk group(DID bank) within the LATA and identified as voice only. In the case ofa call between a UBS provisioned station and a UBR provisioned trunkgroup/DID bank, the UBR rating will apply.

[0210] The following rules apply to usage billing suppression: With UBS,the customer can expect to receive a monthly flat rate charge forintraLATA calling between centrex stations in lieu of per call billing.This monthly flat rate charge is expected to reduce message and tollcharges for these particular calls. Usage billing suppression isavailable between centrex stations provisioned with a custom dialingplan, such as MDP, SDP, or both, and which are within the same LATA. Inone embodiment, voice only calls includes fax calls and non-IntegratedServices Digital Network (ISDN) data transmission at speeds of up to 56KB/second.

[0211] UBS is not “free” usage, but rather is considered flat ratepredictable usage between intraLATA, non-co-located centrex stationsbased on call volume averages on these types of calls. Usage billingsuppression is a subscription option for the whole customer group, i.e., if one centrex telephone number of a customer subscribes to UBS, allcentrex telephone numbers for the customer will have UBS.

[0212] When usage billing suppression is active for a customer group anda triggering centrex telephone number calls an intraLATA centrextelephone number, whose originating calls will trigger, the usage AMArecord (the basis for billing) is suppressed, i.e., discarded formeasured, local calls and for intraLATA toll calls. When a triggeringcentrex telephone number calls an interLATA centrex telephone number,whose originating calls would trigger, the usage AMA record is createdas normal for interLATA toll calls. When a triggering centrex telephonenumber calls a terminating only telephone number, intraLATA orinterLATA, the usage AMA record is created as normal for all calls. Adialed telephone number must be in the dialing plan in order to receiveUBS.

[0213] Usage billing reduction (UBR) is an option for the triggering PBXstations/trunk groups (DID banks) in a subscribing customer group. UBRis available between all provisioned stations whether they are centrexor PBX as long as they are provisioned with usage billing suppression oncentrex and usage billing reduction on PBX and the call is intraLATA. Inorder to be eligible for usage billing reduction the calling station hasto be a fully participating station/trunk group (DID bank) within theLATA and identified as voice only. In the case of a call between a UBSprovisioned station and a UBR provisioned stations/trunk groups (DIDbanks), the UBR rating will apply. In all cases, all pricing paradigmsmust be supportable by billing systems. It should be understood thatsome set up time for billing systems is required each time a change inrating is requested.

[0214] Usage billing reduction allows customers to take advantage ofdiscount usage between intraLATA, inter-centrex/PBX locations. Thefollowing rules apply to usage billing reduction: With UBR, thecustomer's account team can identify and negotiate usage prices based onknown location dialing patterns. These calls are intraLATA callingbetween centrex/PBX fully participating stations/trunk groups (DIDbanks) in lieu of per call billing. Usage billing reduction is availablebetween centrex stations and/or PBX /trunk groups (DID banks) when theyare fully participating in a custom dialing plan, such as MDP, SDP, andwithin the same LATA. The calls must also be voice only. In oneembodiment, voice calls include fax calls and non-ISDN data transfer by56 KB/second (and slower) modems. UBR unlike UBS is not flat ratepricing based on call volume averages, but is a device to allowcustomers to identify location usage patterns and rate according tonegotiated prices, taking into consideration a full range of networkcosts and traffic.

[0215] In one embodiment, the customer selects UBS on centrex and UBR onPBX. In this embodiment, the account team has the option to discountusage on PBX to centrex and centrex to PBX, but the lowestclassification is charged for the usage, i.e., the highest availableusage charge. The lowest classification is also charged for the usagewhen UBR prices are discounted lower for some types of calls, i.e., thehighest available usage charge applies.

[0216] The present invention requires billing systems that are able totranslate originating and destination calling locations/classes ofservice at those locations and are able to rate the usage differently.

[0217] When a triggering centrex or PBX telephone number, whoseoriginating calls would trigger, calls an interLATA centrex or PBXtelephone number, the usage billing record is created as normal forinterLATA toll calls. When a triggering centrex or PBX telephone numbercalls a terminating only telephone number, intraLATA or interLATA, theusage billing record is also created as normal for all calls.

[0218] As discussed above, certain switches are not always configured toprovide the features of the present invention, for example, the 1AESSswitches. Thus, if the customer has centrex or PBX connection serviceout of one of these switches, a trunk side connection is used to connectto an equipped switch. The cost of this private line should be paid forby the customer. The addition of 9+ actually may require two trunkgroups for this embodiment if the access code cannot be sent. Inaddition, local 9+ traffic is sent back to the non-provisioned switch,e.g., the 1AESS, over the PSTN if the non-provisioned switch is the hostswitch for the called local telephone number.

[0219] In order to implement usage billing suppression, a serviceswitching point (SSP) produces AMA records under the same circumstancesthat it normally produces AMA records, i.e., without the abbreviateddialing plan service. When the UBS feature is activated, the servicecontrol point (SCP) produces AMA records with an AIN AMAslpID in themthat the service provider can use to suppress the usage sensitivebilling record. Basically, the AMAslpID is a flag that the serviceprovider can use to discard the AMA record.

[0220] The UBS AMAslpID is sent by the SCP whenever the SCP determinesthat the call is over the PSTN and is intraLATA; and the call is betweentwo triggering centrex telephone numbers. It is noted that the callednumber may not be a terminating only telephone number in the dialingplan. In order to send the AMAsIpID, the call must be a voice call.

[0221] Usage billing suppression and usage billing reduction each havetwo options: ON meaning the AMAslpID will be applied when the call iseligible; and OFF meaning the option was not subscribed to by thecustomer.

[0222] From a call type perspective the following are possiblescenarios:

[0223] 1. Centrex to centrex

[0224] 2. Centrex to PBX

[0225] 3. Centrex to other

[0226] 4. PBX to PBX

[0227] 5. PBX to centrex

[0228] 6. PBX to other

[0229] The UBS results for the two options are as follows: If the UBS ONoption was chosen, only centrex to centrex calls will receive UBStreatment. If the UBS OFF option was chosen, centrex to centrex callswill not receive UBS treatment.

[0230] UBS does have a feature interaction with UBR. If UBS ON is theUBS option and UBR ON is the UBR option: centrex to centrex calls willreceive UBS treatment. Centrex to PBX calls; PBX to PBX calls; and PBXto centrex calls will receive UBR treatment. If UBS OFF is the UBSoption and UBR ON is the UBR option: PBX to PBX calls will receive UBRtreatment. Centrex to centrex calls, centrex to PBX calls, and PBX tocentrex calls will receive normal billing.

[0231] According to one embodiment, usage billing suppression and usagebilling reduction are charged on an overall basis. Thus, everytriggering centrex telephone number must have data entries to supportthe operation of UBS. Consequently, charging on a per triggering centrextelephone number basis achieves the overall basis.

[0232] UBR requires a discount to be applied to the normal usage chargesfor measured local or toll calls. This discount may be different foreach of the two options in this feature, but the discount applies to alltelephone numbers covered by each option in effect. UBR's price for thisreduced usage billing is incremental or vertical on the basic triggeringtelephone number price.

[0233] According to an aspect of the present invention, the SSP producesAMA records under the same circumstances that produces AMA records as ifthe abbreviated dialing plan service did not exist. The service with theoptional UBR feature produces AMA records with an AIN AMAsIpID in themthat the service provider can use to discount the usage sensitivebilling record. Basically, the AMAslpID is a flag that the serviceprovider can use to discount the AMA record after rating it. Thediscount rate comes from the customer's record.

[0234] The UBR AMAslpID is to be sent by the SCP whenever the SCPdetermines: The call is PSTN intraLATA, and the call is between two PBXtelephone numbers that are triggering PBX telephone numbers. The callmust also be a voice call. It is noted that the called telephone numbermay not be a terminating only telephone number in the dialing plan. AnAMAslpID is not sent from the SCP when usage billing treatment is notsubscribed to.

[0235] The UBR results for the two options (UBR ON and UBR OFF) are asfollows: If the UBR ON option was chosen, only PBX to PBX calls will getthe UBR AMAslpID for PBX. If the UBR OFF option was chosen, PBX to PBXcalls will not get the UBR AMAsIpID for PBX. Centrex to centrex calls,centrex to other calls, and PBX to other calls are never eligible forUBR.

[0236] UBR has an additional feature interaction with UBS. If UBR OFF isthe UBR option and UBS ON is the UBS option, centrex to PBX calls, PBXto PBX calls, and PBX to centrex calls will get normal billing. Centrexto centrex calls will receive UBS.

[0237] According to the present invention, it is important to identifythe calling party ID with an indicator so that it is known whether thecalling party is a PBX or centrex station. UBS and UBR are features thatallow a specific customer to subscribe to them by general class ofservice, i.e., centrex or PBX. A specific customer may or may notsubscribe to UBS for centrex and may or may not subscribe to UBR. Thebilling option taken for centrex has no impact on the billing optionthat may be selected for PBX and vice versa.

[0238] Even if a specific customer subscribes to UBS or UBR, there areseveral conditions that must be met before either may be applied to acall. Each condition requires that information relative to thatcondition be derived from call processing. The most fundamental questionis what is the general class of service of the calling party: centrex orPBX? The second fundamental question is does the customer subscribe tobilling for that class of service, i.e., is the UBS option ON or is theUBR option ON? From there, the other conditions, e. g., the billing forthe called class of service, intraLATA, and voice are factors that mustbe derived from call processing.

[0239] Referring now to FIG. 9, using the value of the call type code,it is determined whether the value of the call type code is not 0 atstep s3000. If the call type code value is not 0, primary billing logicwill be used because a value for the call type code was previouslydetermined, either as part of OCS or ASF-RTL. If the value of the calltype code is 0 (default-never determined) (step s3000: NO), alternatebilling logic will be used because a value of the call type code was notdetermined previously.

[0240] At step s3002, it is determined whether the value of the calltype code indicates that the call is intraLATA, centrex to centrex (avalue of 111 in one embodiment) and the value of the ASF result is 0(i.e., ASF-RTL has not changed the routing). If both values are 111 and0, respectively, the service logic will determine whether the call is avoice call at step s3004. If the call is determined to be a voice call,at step s3006 the AMA billing record is set so that UBS applies. If thecall is determined not to be a voice call, at step s3008 the value ofthe AMA billing record is not set such that normal SSP billing willapply.

[0241] If the call type code is not 111 or the ASF result is 0, at steps3010 it is determined whether the call type code indicates that thecall is intraLATA, centrex to PBX (a value of 112 in an embodiment) thevalue of the ASF result is 0 (i.e., ASF has not changed the routing). Ifboth values are 112 and 0, respectively, the service logic willdetermine whether the call is a voice call at step s3004. If the call isdetermined to be a voice call, at step s3006 the AMA billing record isset so that UBS applies. If the call is determined not to be a voicecall, at step s3008 the value of the AMA billing record is not set suchthat normal SSP billing will apply.

[0242] If the call type code is not 112 or the ASF result is not 0, atstep s3012 it is determined whether the call type code indicates thatthe call is intraLATA, PBX to centrex (a value of 121 in a preferredembodiment) and the value of the ASF result is 0 (i.e., ASF has notchanged the routing). If both values are 121 and 0, respectively, thecommon service logic CPR will determine whether the call is a voice callat step s3004. If the call is determined to be a voice call, at steps3006 the AMA billing record is set so that UBS applies. If the call isdetermined not to be a voice call, at step s3008 the value of the AMAbilling record is not set such that normal SSP billing will apply.

[0243] If the call type code is not 121 or the ASF result is not 0, atstep s3014 it is determined whether the call type code indicates thatthe call is intraLATA, PBX to PBX (a value of 122 in a preferredembodiment) and the value of the ASF result is 0 (i.e., ASF has notchanged the routing). If both values are 122 and 0, respectively, thecommon service logic CPR will determine whether the call is a voice callat step s3004. If the call is determined to be a voice call, at steps3006 the AMA billing record is set so that UBS applies. If the call isdetermined not to be a voice call, at step s3008 the value of the AMAbilling record is not set such that normal SSP billing will apply.

[0244] If the call type code is not 122 or the ASF result is not 0, atstep s3016 it is known that the call type code is not 111, 112, 121 or122, or the ASF result is not 0. Thus, the value of the AMA billingrecord is not set such that normal SSP billing will apply at step s3008.

[0245] Returning to step s3000, when it is determined that the call typecode is 0 (in other words, the call type code has not yet beendetermined), the service logic determines whether the calling party is acentrex or PBX station at step s3018. Subsequently, at step s3020 it isdetermined whether the called party is a centrex or PBX station. If thecalling party is a centrex or PBX station and the called party is acentrex or PBX station, as determined at step s3022, at step s3024 it isdetermined whether the call is intraLATA. If either the calling party isnot a centrex or PBX station or the called party is not a centrex or PBXstation, at step s3008 the value of the AMA billing record is not setsuch that normal SSP billing will apply.

[0246] If the call is determined to be intraLATA at step s3024, at steps3026 the call type code is set in accordance with the results obtainedin steps s3018 and s3020. Then, the logic starting at s3002 repeats, asdescribed above, with the newly set call type code. If the call isdetermined not to be an intraLATA call, at step s3008 the value of theAMA billing record is not set such that normal SSP billing will apply.

[0247] The billing department of the local exchange carrier processesUBS AMA records such that the customer does not receive local usagecharges for the call. The monthly recurring charges for the UBS featurerecover costs in lieu of local usage billing. The billing data can becaptured for financial verification to ensure that costs are met. In analternate embodiment, the UBS AMA records are sent to another localservice provider if the subscriber's service is provided via resale orunbundled. Originating centrex calls that are intraLATA are alsoprocessed for originating access billing. The billing department of thelocal exchange carrier processes the UBR AMA records such that customerreceives the discounted PBX local usage charge for the call. In thiscase, the monthly recurring charge for the UBR-PBX feature and thediscounted rate are how the cost is recovered in lieu of normal localusage billing. The billing data is captured to bill the discounted PBXlocal usage charge and for financial verification that costs are met. Inan alternate embodiment, the UBR AMA records are sent to another localservice provider if the subscriber's service is provided via resale orunbundled. Originating centrex calls that are intraLATA are alsoprocessed for originating access billing.

[0248] In one embodiment, an additional type of billing treatment isprovided. This additional type of billing treatment can be subscribed toby the customer if the customer subscribes to ASF-RTL. This additionalbilling treatment will be referred to as remote tie line billingreduction (RBR) (step s210 of FIG. 2). In this embodiment, only usagebilling reduction (rather than suppression) is available.

[0249] RBR is independent from UBR and UBS because it is possible tosubscribe to RBR without subscribing to UBS or UBR. When a customersubscribes to RBR, RBR applies to all ASF-RTL telephone numbers. Inother words, it is not possible for some ASF-RTL telephone numbers to betreated according to RBR while other ASF-RTL telephone numbers are nottreated in accordance with RBR. In addition, with RBR the general classof service originating the call, i.e., centrex or PBX, is irrelevant. Ifthe calling party places a voice call that is routed to an ASF-RTLtelephone number in the same LATA as the calling party, and if RBR isactive, the call will be billed in accordance with RBR. Otherwise,normal switch billing occurs.

[0250] RBR is applied on the PSIN leg of the call from the originatingnumber to the ASF-RTL host switch, where the call is passed over toprivate facilities. RBR can be discounted to any level including zerorating based on an RBR flat rate price. In one embodiment, there isflexibility within the RBR rating so that charges can be differentiatedbased on class of service and location routing through the ASF-RTL hostswitch, but all pricing paradigms must be supportable by billingsystems. The present invention only supplies the AMAslpID and class ofservice. Location identification comes from the normal billing process.RBR is available and applicable only when all ASF-RTL host switches arefully participating.

[0251] RBR is also a variant of UBS and is similar to UBR. RBR appliesonly to remote tie lines (RTLs) in the ASF-RTL feature. RBR says that adiscount will be applied to the normal usage charges for measured localor toll if the calling party number and remote tie line telephone numberare in the same LATA regardless of what the true called number is.Hence, the first leg, the PSTN leg, is discounted if the calling partynumber and remote tie line telephone number are in the same LATA. It isnoted that there is no requirement for a remote tie line telephonenumber to be in the same LATA as the switches that originate traffic tothe remote tie line telephone number.

[0252] RBR exists for all RTL telephone numbers or it exists for none.Unlike UBR, RBR may range from a 0% (not logical) to a 100% (effectivelyUBS for remote tie line telephone numbers) discount. The discount, ifnot the same for all remote tie line telephone numbers, may vary bylocations, and if it does, the rating systems in the billingorganization will be responsible for applying such location sensitivediscounts. RBR's price for this reduced usage billing is incremental orvertical on the basic remote tie line telephone number price. Anadvantage of RBR is that the customer receives a more predictablemonthly bill instead of a bill that potentially fluctuates more widely.

[0253] Again, the present invention does nothing with respect to AMAbilling without the UBR, UBS, or RBR feature. The switch produces AMArecords under the same circumstances that it would produce AMA recordsas if the present invention did not exist. The present invention withthe optional RBR feature produces AMA records with an AIN AMAslpID inthem that a finance department can use to discount the usage sensitivebilling record. Basically, the AMAslpID is a flag that the financedepartment can use to discount the AMA record after rating it. Thediscount rate will come from the customer's record in the financedepartment. Unlike UBS and UBR, RBR does not have any options.

[0254] An RBR AMAsIpID is to be sent by the SCP whenever the SCPdetermines: The call is PSTN intraLATA; the call is between a triggeringtelephone number and a remote tie line telephone number; and the call isa voice call.

[0255] From a call type perspective the following are possiblescenarios:

[0256] 1. Centrex to remote tie line telephone number (intraLATA, voice)

[0257] 2. Centrex to remote tie line telephone number (intraLATA, data)

[0258] 3. Centrex to remote tie line telephone number (interLATA, voiceor data)

[0259] 4. PBX to remote tie line telephone number (intraLATA, voice)

[0260] 5. PBX to remote tie line telephone number (intraLATA, data)

[0261] 6. PBX to remote tie line telephone number (interLATA, voice ordata) The RBR results for the six scenarios are as follows: Scenarios #1and #4 would get the RBR AMAsIpID. Scenarios #2, #3, #5 and #6 are nevereligible for RBR.

[0262] RBR does have a feature interaction with UBS or UBR. If RBRexists for a call, UBS and UBR can not exist for the same call. If UBSor UBR exists for a call, RBR can not exist for the same call.

[0263] From a practical viewpoint, RBR is charged on an overall basis.If RBR is employed, every triggering telephone number, centrex and PBX,is assumed to have RBR such that data entries to support the operationof RBR are not necessary. Charging on a per remote tie line telephonenumber basis or on a per triggering telephone number, centrex and PBXbasis are ways to achieve the overall basis.

[0264] The billing department of the local exchange carrier processesthe RBR AMA records such that the customer receives the discounted RBRlocal usage charge for the call. It is noted that the monthly recurringcharge for the RBR feature and the discounted rate recover costs in lieuof normal local usage billing. The billing data is captured to bill thediscounted RBR local usage charge and for financial verification thatcosts are met. In an alternate embodiment, the RBR AMA records are sentto another local service provider if the subscriber's service isprovided via resale or unbundled. Originating centrex calls that areintraLATA are also processed for originating access billing.

Virtual SMDR

[0265] According to another embodiment of the present invention, avirtual station message detail recording (Virtual SMDR) system isprovided. Virtual SMDR is an automated call data collection and reportgenerating system that collects call data at a centralized servicecontrol point (SCP). The system formats the call data at a Virtual SMDRfront end server as SMDR data and generates a call report based on theSMDR data at a host central processing unit.

[0266] The invention consolidates the data collection process for reportgeneration at an SCP, as opposed to individual switches, and relies onsampling data routinely available to the SCP from the switches. VirtualSMDR therefore provides several advantages over the conventional SMDRsystems. First, the customer does not need to purchase and load SMDRsoftware packages at every service switching point that handles calls toand from the private telecommunications network. Likewise, the customerdoes not incur the expense of each switch's required connectivity withthe host so that the SMDR data is available to the customer. Rather,only one connection is necessary: between the SCP and the host. Also,because the SCP does not require actual SMDR data generated by theswitches, but only needs a sampling of SMDR-like data, Virtual SMDR hassufficient flexibility to generate calling records for PBX calls, aswell as centrex calls, handled by the SCP.

[0267] This present invention upgrades the current SMDR data collectionand report generation system, simplifying the process, yet increasingfunctionality. FIG. 10a illustrates portions of an exemplarytelecommunications network in association with one embodiment of thepresent invention for implementing the Virtual SMDR reporting system.The network includes service switching points (SSP) 4000 and 4020, andan Integrated Service Control Point (ISCP) 4002 in a public switchedtelecommunications network (PSTN). The PSTN also includes a local signaltransfer point (STP) 4014 and a regional STP 4013 for directingsignaling communications to the ISCP 4002.

[0268] The information generated by the PSTN is provided by way of adata distributor interface node 4004. The data is then transmitted to aVirtual SMDR front end server server 4010, such as a CentrexSMART frontend (CFE) server. From the front end server 4010, the data is sent to aVirtual SMDR host central processing unit 4012, such as a CentrexSMARThost. The data distributor 4004 serves as an interface between the PSTNand the network that formats and generates the Virtual SMDR reports. Thedata distributor 4004 has more than one interface type, including abilling interface and an American Standard Code for InformationInterchange (ASCII) interface, for communicating with the ISCP 4002. Thepreferred interface for sampling is the data distributor ASCIIinterface, which is depicted in FIG. 10a. In an embodiment of theinvention, the data distributor 4004 is implemented according toBellcore SR 3918, ISCP Data Distributor/ASCII Collection SystemInterface Specification, available from Telecordia, Murray Hill, N.J.,the contents of which are expressly incorporated by reference herein inits entirety.

[0269] In FIG. 10a, the SCP is an Integrated SCP, implemented with aBellcore Integrated Service Control Point and loaded with ISCP softwareVersion 4.4 (or higher), available from Telecordia, Murray Hill, N.J.Any compatible SCP may be incorporated in the invention. For example,FIG. 10b is a block diagram of the system incorporating a Lucent SCP(LSCP) 4018, with software release 94, or higher, available from LucentTechnologies, Inc., in place of ISCP 4002. As indicated by FIG. 10b, theincorporation of the LSCP 4018 eliminates the requirement for the datadistributor 4004, as discussed further below. The remaining systems forimplementing Virtual SMDR with the LSCP 4018, as shown in FIG 10 b, areidentical to those systems used in conjunction with the ISCP 4002.

[0270]FIG. 11 is a flowchart showing exemplary steps for implementingVirtual SMDR according to the system depicted in FIG. 10a. At steps4160, the ISCP 4002 receives a query, via the STPs 4013 and 4014, fromthe originating SSP 4000. The call is processed in a known fashion,including sending call processing instructions to the originating SSP4000 to direct the call, for example, to a terminating SSP 4020, andreceiving and processing a query from the terminating SSP 4020.

[0271] For purposes of Virtual SMDR, receiving queries from the variousswitches is significant in that the queries provide the callinginformation from which the ISCP 4002 samples the SMDR-like data. Inother words, the switches generate the basic data for callidentification and handling as they would for any call employing theabbreviated dialing plan system of the present invention. The data issampled at step s4162 by the SCP 4002. In particular, the ISCP 4002selects data similar to the SMDR data that would be provided directlyfrom the SSPs to an SMDR host in a conventional system. For instance,the ISCP 4002 samples the calling party ID, the called party ID, acustomer group ID, the start time, and the end time.

[0272] Virtual SMDR is compatible with other features of the presentinvention. For example, for calls routed according to the automaticselection of facilities-remote tie line (ASF-RTL) feature, an additionalvariable indicating the original called party ID is included in thesampling. The ISCP 4002 and ultimately the Virtual SMDR host 4012 canthen distinguish between the called party ID (i.e., the telephone numberhosting a private trunk group for selected calls) and the telephonenumber of the desired destination of the caller.

[0273] After sampling the SMDR-like data from the originating SSP 4000and the terminating SSP 4020, the ISCP 4002 outputs the data to the datadistributor ASCII interface 4004 at step 4164. In the embodiment of theinvention shown in FIG. 10a, the ISCP 4002 does not process or store thedata for purposes of Virtual SMDR, but rather forwards the data withoutmodifying the data. The data distributor 4004 stores and sorts the dataat step s4166.

[0274] At step s4164, the data distributor 4004 produces two callrecords for any one telephone call made by a customer. The first recordis an initial call or attempt data record created whenever the call isplaced. The second record is a completion data record, which indicatesthe final status of the telephone call. The two records have a commonidentifier called, in an embodiment incorporating the ISCP 4002, “echo”data. The echo data enables the front end server 4010 to associate theattempt data record and the corresponding completion data record into aconsolidated SMDR call record.

[0275] At step s4168 the data is sent from the data distributor 4004 tothe front end server 4010 pursuant to known file transfer protocol (FTP)methods. In an embodiment of the invention, the data distributor 4004transmits the call data at an interval of once per hour. At this point,the data is still not associated with SMDR processing and is simply acollection of telecommunications data as sampled by the ISCP 4002. Thefront end server 4010 processes the sampled data into an SMDR format atstep s4170. The SMDR data is then essentially equivalent, from aformatting standpoint, to SMDR data provided directly from the switchesin a conventional SMDR systems. The processing performed by the frontend server 4010 includes associating the two sampled data records fromthe data distributor 4004, using the echo data identifier, so that theentire telephone call has a single SMDR record. The front end server44010 then stores the formatted SMDR data at step s4172 and sends it tothe Virtual SMDR host processor 4012 at step s4174.

[0276] The host processor 4012 stores the data in an SMDR format foraccess by the customer. The SMDR data is stored in a table format foreach call, in one embodiment of the invention. The customer 4022 is ableto request a variety of data combinations, from a full report toisolated call variables, depending on the customer's needs. The customer4022 requests a Virtual SMDR report at step s4178 to be sent via anetwork interface using known connectivity techniques to anadministrative facility (not pictured). The host processor 4012generates the report at step s4176 according to the parameters requestedby the customer, drawing from the data table associated with thetelephone calls, and forwards the Virtual SMDR reports.

[0277] In a second embodiment, the SCP is an LSCP 4018, shown in FIG.10b. The LSCP 4018 receives and stores call service data from thevarious originating and terminating switches, such as originating SSP4000 and terminating SSP 4014, as in the first embodiment. However,there is no need for a separate interfacing system, such as the datadistributor 4004 of FIG. 10a. Instead, the LSCP 4018 accomplishes allprocessing necessary for the front end server 4010 to associate recordsand to format the calling data into SMDR data. Therefore, according tothe second embodiment of the invention, sampling the call data at steps4162, storing the call data at step s4164 (in this case in the LSCP4018 rather than in the data distributor 4004) and sending the call datato the front end server at step s4168 are all performed by the LSCP4018. Step s4164, sending data to the data distributor 4004, is thuseliminated. As in the first embodiment, two call records are generated,the initial call or attempt data and the completion data. These two callrecords are sent from the LSCP 4018 to the front end server 4010 at steps4168 along with the other sampled call data. The two records have acommon identifier, called a “notification ID” data in this embodiment,that enables the front end server 4010 to associate the attempt datarecord and the corresponding completion data record to create and storean SMDR formatted table at steps s4170 and s4172 of FIG. 11. The reportgenerated by the host processor 4012, which is accessible by thecustomer 4022, is the same regardless of the type of SCP used in theVirtual SMDR system.

[0278] Although the invention has been described with reference toseveral exemplary embodiments, it is understood that the words that havebeen used are words of description and illustration, rather than wordsof limitation. Changes may be made within the purview of the appendedclaims, as presently stated and as amended, without departing from thescope and spirit of the invention in its aspects. Although the inventionhas been described with reference to particular means, materials andembodiments, the invention is not intended to be limited to theparticulars disclosed; rather, the invention extends to all functionallyequivalent structures, methods, and uses such as are within the scope ofthe appended claims.

[0279] In accordance with various embodiments of the present invention,the methods described herein are intended for operation as softwareprograms running on a computer processor. Dedicated hardwareimplementations including, but not limited to, application specificintegrated circuits, programmable logic arrays and other hardwaredevices can likewise be constructed to implement the methods describedherein. Furthermore, alternative software implementations including, butnot limited to, distributed processing or component/object distributedprocessing, parallel processing, or virtual machine processing can alsobe constructed to implement the methods described herein.

[0280] It should also be noted that the software implementations of thepresent invention as described herein are optionally stored on atangible storage medium, such as: a magnetic medium such as a disk ortape; a magneto-optical or optical medium such as a disk; or a solidstate medium such as a memory card or other package that houses one ormore read-only (non-volatile) memories, random access memories, or otherre-writable (volatile) memories. A digital file attachment to email orother self-contained information archive or set of archives isconsidered a distribution medium equivalent to a tangible storagemedium. Accordingly, the invention is considered to include a tangiblestorage medium or distribution medium, as listed herein and includingart-recognized equivalents and successor media, in which the softwareimplementations herein are stored.

[0281] Although the present specification describes components andfunctions implemented in the embodiments with reference to particularstandards and protocols, the invention is not limited to such standardsand protocols. Each of the standards for Internet and other packetswitched network transmission (e.g., TCP/IP, UDP/IP, HTML, SHTML, DHTML,XML, PPP, FTP, SMTP, MIME); peripheral control (IrDA; RS232C; USB; ISA;ExCA; PCMCIA), and public telephone networks (ISDN, ATM, XDSL) representexamples of the state of the art. Such standards are periodicallysuperseded by faster or more efficient equivalents having essentiallythe same functions. Accordingly, replacement standards and protocolshaving the same functions are considered equivalents.

What is claimed is:
 1. A method for efficiently routing a call from aprivate telecommunications network through a public switched telephonenetwork (PSTN) and a plurality of trunk groups, including at least oneprivate trunk group, the method comprising: setting a called party ID tocorrespond to an automatic selection of facilities-remote tie-line(ASF-RTL) telephone number; routing the call from a first switch to aswitch hosting the ASF-RTL telephone number via a public trunk group;and routing the call from the switch hosting the ASF-RTL telephonenumber to a private customer facility hosting the called party numbervia a private trunk group.
 2. The method for efficiently routing a callfrom a private telecommunications network according to claim 1, furthercomprising: initially determining whether the call is subject to ASF-RTLprocessing.
 3. The method for efficiently routing a call from a privatetelecommunications network according to claim 1, further comprisingrouting the call from the switch hosting the ASF-RTL telephone number tothe private customer facility hosting the called party number via asecond trunk group whenever the private trunk group is unavailable. 4.The method for efficiently routing a call from a privatetelecommunications network according to claim 1, further comprisingrouting the call from the private customer facility to an internationalgateway when a service control point determines that a called partynumber of the call is an international number.
 5. The method forefficiently routing a call from a private telecommunications networkaccording to claim 1, wherein the private telecommunications networkcomprises at least one of a central exchange service and a privatebranch exchange.
 6. A method for efficiently routing a call to a calledparty number from a private telecommunications network through a publicswitched telephone network (PSTN) and a plurality of trunk groups,including at least one private trunk group, the method comprising:querying a service control point for call processing instructions forthe call, the query comprising a calling party ID corresponding to acalling party number and a called party ID corresponding to the calledparty number; identifying whether the call is subject to automaticselection of facilities-remote tie-line (ASF-RTL) processing; when thecall is subject to ASF-RTL processing, setting the called party ID tocorrespond to an ASF-RTL telephone number and setting an original calledparty ID to correspond to the called party number; transmitting from theservice control point to an originating service switching point callprocessing instructions, the call processing instructions comprising thecalling party ID, the called party ID and the original called party ID;routing the call to a host service switching point corresponding to thecalled party ID; querying the service control point for additional callprocessing instructions, the query comprising the calling party ID, thecalled party ID and the original called party ID; determining additionalcall processing instructions, comprising at least one private trunkgroup through which to route the call, and resetting the called party IDto correspond to the original called party ID; transmitting theadditional call processing instructions from the service control pointto the host service switching point; and routing the call from the hostservice switching point to a private switching facility of the customervia the at least one private trunk group.
 7. The method for efficientlyrouting a call to a called party number from a privatetelecommunications network according to claim 6, further comprising:routing the call to a destination service switching point; andconnecting the calling party number with the called party number.
 8. Themethod for efficiently routing a call to a called party number from aprivate telecommunications network according to claim 6, in which thedetermining the additional call processing instructions furthercomprises: determining at least one alternative trunk group and at leastone carrier by which to route the call; and in which the routing furthercomprises: determining whether the at least one private trunk group isfull or unavailable; and whenever the private trunk group is determinedto be full or unavailable, routing the call to the private switchingfacility via the at the least one alternative trunk group and, when theat least one alternative trunk group is a public trunk group, the atleast one carrier.
 9. The method for efficiently routing a call to acalled party number from a private telecommunications network accordingto claim 6, wherein determining whether the telephone call from thecalling party number to the called party number is subject to remotetie-line processing is based on a type of call.
 10. The method forefficiently routing a call to a called party number from a privatetelecommunications network according to claim 6, wherein determiningwhether the telephone call from the calling party number to the calledparty number is subject to remote tie-line processing is based on atleast one of a type of call, a date, a day of week, a time of day and apercentage allocation.
 11. The method for efficiently routing a call toa called party number from a private telecommunications networkaccording to claim 6, wherein the private switching facility comprisesan electronic tandem network node.
 12. The method for efficientlyrouting a call to a called party number from a privatetelecommunications network according to claim 6, wherein the privateswitching facility comprises an asynchronous transfer mode switch. 13.The method for efficiently routing a call to a called party number froma private telecommunications network according to claim 6, wherein theprivate telecommunications network comprises at least one of a centralexchange service and a private branch exchange.
 14. A system forefficiently routing a call from a private telecommunications networkthrough a public switched telephone network (PSTN) and a plurality oftrunk groups, including at least one private trunk group associated withan automatic selection of facilities-remote tie-line (ASF-RTL) telephonenumber, the system comprising: a service control point that processestelephone calls; a first service switching point that queries saidservice control point in response to the call from the privatetelecommunications network; a second service switching point comprisinga host switching point for the at least one private trunk associatedwith the ASF-RTL telephone number; and a third service switching pointcomprising a private facility of the customer; wherein said servicecontrol point determines from the query whether the call is subject torouting through said second switching point, and if so, instructs saidfirst service switching point to route the call to said second serviceswitching point, and wherein said second service switching point routesthe call to said third service switching point via the private trunkgroup associated with the ASF-RTL telephone number.
 15. The system forefficiently routing a call to a called party number from a privatetelecommunications network according to claim 14, further comprising aninternational gateway, wherein when said service control pointdetermines that the called party number is an international telephonenumber, the call is routed from said third service switching point tosaid international gateway.
 16. The system for efficiently routing acall to a called party number from a private telecommunications networkaccording to claim 14, further comprising an interexchange carrier,wherein when said service control point determines that the called partynumber is a long distance telephone number, the call is routed from saidthird service switching point to said interexchange carrier.
 17. Thesystem for efficiently routing a call from a private telecommunicationsnetwork according to claim 14, wherein the private telecommunicationsnetwork comprises at least one of a central exchange service and aprivate branch exchange.
 18. A system for efficiently routing a callfrom a private telecommunications network of a customer through a publicswitched telephone network (PSTN) and a plurality of trunk groups,including at least one private trunk group, the system comprising: aservice control point that processes telephone calls, said servicecontrol point comprising an automatic selection of facilities-remotetie-line (ASF-RTL) routing database; an originating service switchingpoint that launches a trigger to said service control point in responseto a call initiated at a telephone in the private telecommunicationsnetwork of the customer; a host switching facility that hosts an ASF-RTLtelephone number associated with the at least one private trunk group,said host switching facility receiving the call from said originatingservice switching point based on instructions from said service controlpoint; and a private switching facility that hosts at least a secondprivate telecommunications network of the customer, said privateswitching facility connecting to said host switching facility by atleast the private trunk group and receiving the call from said hostswitching facility based on routing instructions from said servicecontrol point; wherein said service control point receives a callingparty ID and a called party ID corresponding to the call from saidoriginating service switching point and determines from the ASF-RTLrouting database whether the call is subject to ASF-RTL routing;wherein, if the call is subject to ASF-RTL routing, said service controlpoint sets the called party ID to be the ASF-RTL telephone numberassociated with the at least one private trunk group so that saidoriginating service switching point forwards the call to said hostswitching facility; and wherein the routing instructions comprisetransmitting the call from said host switching facility to said privateswitching facility via the at least one private trunk group.
 19. Thesystem for efficiently routing a call to a called party number from aprivate telecommunications network according to claim 18, wherein therouting instructions comprise the plurality of trunk groups, includingthe at least one private trunk group and a plurality of alternativetrunk groups, and a plurality of carriers; and wherein said hostswitching facility selects one of the plurality of alternative trunkgroups and, when the selected alternative trunk group is a public trunkgroup, one of the plurality of carriers by which to forward the call tosaid private switching facility when the at least one private trunkgroup is not available.
 20. The system for efficiently routing a call toa called party number from a private telecommunications networkaccording to claim 19, wherein each of the plurality of trunk groupscomprises a multi-frequency trunk.
 21. The system for efficientlyrouting a call to a called party number from a privatetelecommunications network according to claim 18, wherein said privateswitching facility comprises an electronic tandem network node.
 22. Thesystem for efficiently routing a call to a called party number from aprivate telecommunications network according to claim 18, wherein saidprivate switching facility comprises an asynchronous transfer modeswitch.
 23. The system for efficiently routing a call to a called partynumber from a private telecommunications network according to claim 18,wherein the ASF-RTL routing database comprises routing information basedon at least one of a type of call, a date, a day of week, a time of dayand a percentage allocation.
 24. The system for efficiently routing acall to a called party number from a private telecommunications networkaccording to claim 18, wherein the private telecommunications network ofthe customer comprises at least one of a central exchange service and aprivate branch exchange.
 25. The system for efficiently routing a callto a called party number from a private telecommunications networkaccording to claim 18, further comprising a terminating serviceswitching point to which the call is directed from said privateswitching facility, wherein said service control point instructs saidterminating service switching point to terminate the call at the calledparty number.